4 2 6 MeetIP Manual Administracion MTv3.8.5 Eng V1.2

Meet IP
MT 3.8.5 Getting Started
Contents
• 1 System Requirements
• 2 Instalation Guide
• 3 Setup Wizard
o 3.1 EULA
o 3.2 Administrator Details
o 3.3 Server Details (CD ROM and
USB version)
o 3.4 Server Details (Vserver version)
o 3.5 Licensing
o 3.6 Locality
o 3.7 Music On Hold
o 3.8 Trunk
o 3.9 Confirm
o 3.10 Logging into the system
o 3.11 Updates
o 3.12 License Updates
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System Requirements
What is needed to get the PBXware up and running?
Server Hardware
• Standard x86-compatible server hardware

• 1.6 GHz or faster CPU minimum (2.6 GHz or more recommended)
• Compatible processors include:
Intel®: Celeron®, Pentium® II, Pentium III, Pentium 4, Xeon™, Core® series,
Core® 2, Core® iX, Atom
AMD™: Athlon™, Athlon MP, Athlon XP, Athlon X series, Duron™,
Opteron™
Via: C3, C7
Memory
• 1 GB minimum (2 GB or more recommended)
Display
• None required. (Exception is during the installation only for)
Disk Drives
• Free space for installation:

• 15 GB minimum for CD ROM/USB Installation Method
• For PBXware in Vserver environment, please check disk space requirements for
Bicom Systems SERVERware
• Standard CD-ROM/DVD-ROM drive
Local Area Networking
• Any Ethernet controller supported by the operating system

• Network configured and fully setup with DHCP service
• Optional: If the server is to operate from a private LAN IP then the firewall must
be opened to the following ports: TCP: 80, 81, 443, 10001, 5038, 5060-5069,
UDP: 4569, 5060-5069, 10000-20000
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Supported Browsers
System can be administered by using one of the following web browsers:
• Internet explorer 7.0+

• Firefox 3.5+
• Opera 10.*+
• Safari 4.0+
• Chrome 9+
Support Requirements
In order to provide system support we need the access to the system server by SSH,
HTTP/HTTPS protocols, following is required:
• Access to system server as user 'root'

• Networking setup and fully configured to port forward (or firewall opened) to
ports 2020, 80, 81 and 443.
With above in place our technicians will be able to troubleshoot issues. We regret that
we are not able to support systems, that do not satisfy above requirements. We ask for
understanding.
If not sure how to install PBXware we offer a professional installation service. Please
contact sales or visit our web site for full details.
Instalation Guide
The PBXware can be installed by one of the following installation methods:
CD ROM installation method is used to install the PBXware onto a commodity

PC/server hardware. The installation process installs a Linux operating system,
PBXware and all other necessary applications onto the system hard drive. Installation is
easy, fast and includes everything needed to successfully install and operate.
WARNING:
CDROM will install the PBXware on hard disk and will erase all existing data,
operating system and other files.
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• USB installation method is similar to CD ROM. Only difference is the boot
medium which holds PBXware installation files.
• VIRTUAL SERVER installation is a PBXware system contained within a

VSERVER (also known as VPS) on Bicom Systems SERVERware installation.
• VMware image is pre-installed with PBXware system so you can test it on your
own computer without the special hardware dedicated for this.
Any of the above formats will start the machine/Vserver/image and all necessary
software allowing administrator to login with browser into this machine/Vserver. The
Administrator can then license the system by entering the licence number or use it in
FREE mode. The system will contact our licensing server for authorization. Upon
successful licensing, systems can be used normally by logging into the system.
CD ROM
To install the system using the CD ROM do the following:
1. Download and burn CD image from http://bicomsystems.com/download

2. Boot up the system, wait for installer to start and follow the instructions
3. The system will be installed and rebooted. Once the system is booted, the
display will show obtained IP address.
4. Please login into setup wizard using your favorite web browser by navigating to
http://$IPADDRESS (For Example: http://192.168.1.2). Default username is
"root". Default password is "pbxware"
IMPORTANT:
• Before continuing please read System Requirements.

• The setup wizard has a security username/password to prevent unauthorized
access. Defaults are: username: root and password: pbxware. The setup wizard
will ask for the password to be changed in one of the setup wizard steps. After it
is changed, It is very important to remember this password since it is the system
root and setup wizard password.
USB
Installing the system from USB is no different than using CD ROM version. Only
difference is how you put downloaded image to the USB:
1. Download USB image from http://bicomsystems.com/download

2. Download PBXware image writer from downloads page
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3. Copy USB image to USB drive using PBXware image writer
4. Set your machines BIOS to boot from USB device
5. Boot the system from USB device
After the image boots from USB device, everything else is the same as the system
booted from CD ROM.
IMPORTANT:
• Before continuing please read System Requirements.

• The setup wizard has a security username/password to prevent unauthorized
access. Defaults are: username: root and password: pbxware. The setup wizard
will ask for the password to be changed in one of the setup wizard steps. After it
is changed, It is very important to remember this password since it is the system
root and setup wizard password.
Vserver installation
Installing PBXware in Vserver environment is only different in the beginning of the

process of installation.
1. Create a Vserver in SERVERware, for which you need to reference the

SERVERware documentation.
2. After Vserver with PBXware is created, go to Setup Wizard and configure your
PBX system.
VMware image
VMware image is used solely for testing purposes and is not supported for deployment.
This means that the user needs to supply the FREE license when asked for, and use the
system only for testing purposes.
1. Download and extract the package to your disk

2. Go to PBXware_3 folder that you get after extracting the image
3. Open PBXware_3.vmx file in VMware Player/Workstation/Fusion
4. Run the virtual machine
Important
Since this is already installed PBXware system inside VMware image, you only need to
go through Setup Wizard for initial configuration of the system and then you will be
able to use it.
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Image is made with bridged networking so the system should get its own IP address
which you can access in any compatible web browser.
Obtaining IP Address
The PBXware IP address can be obtained by:
• Connecting a monitor to the PBXware unit and obtaining the IP from the
command prompt by entering ifconfig command
• Installing the PBXware Finder on a Windows machine which is also connected
to the same network. PBXware Finder can be obtained at:
http://downloads.bicomsystems.com/pbxware_finder/pbxware-finder-1.0.exe
• Obtaining it from the DHCP server
Setup Wizard
Setup Wizard is designed to collect essential data in order to get the system up and
running. After the setup wizard has finished, the system should be fully licensed, fully
operational and ready for use.
In order to login to the setup wizard, please point your browser to:
https://$IPADDRESS:81/ (For example:https://192.168.1.2:81/)
IMPORTANT:
The setup wizard has a security username/password in order to prevent unauthorized

access. The defaults are: username: root and password: pbxware . The setup wizard will
ask for the password to be changed in one of the setup steps. After it is changed, it is
very important to remember this password since it is the system root and the setup
wizard password.
EULA
EULA
7
The EULA (end user license agreement) is the first setup wizard step. Please read the
EULA and type '"I agree"' to proceed.
NOTE:
If you do not agree with the EULA, please remove the installation media, and system
software and return the license issued.
Administrator Details
Provide the details of the user who will administer the system. These values are used
when logging into PBXware
Administrator Details
• E-mail/Confirm E-mail
Provided email address is used as a username for logging in

([a-z] [0-9] [@_.-])
• Password/Confirm Password
Provided password is used for logging in

([a-z] [0-9])
Server Details (CD ROM and USB version)
These are system and network fields necessary for proper system operation.
• Root Password
PBXware prompts for this password during the system/ssh login and when
accessing system services through the interface.
• Confirm Password
Re-type the Root Password entered in the field above
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• Time zone
Select the time zone, in which PBXware is located.
Server Details
• Hostname
The name given to the machine which will identify the system on the network
(ex. "myhost")
([a-z] [0-9])
• Interface
The PBXware interface uses (LAN/WAN). If the PBXware is in the LAN

interface, select it here. In some cases where PBXware is installed on appliances
in WAN mode, select WAN here.
(Select box)
• Use DHCP
Whether PBXware is using DHCP or the static IP address. It is recommended to

always set PBXware on a static IP address, in that case this option should be set
to 'No'.
(Option buttons)
• IP Address
If PBXware is in the LAN, provide its static IP address here.

([0-9])
• Netmask
This field is calculated automaticaly and contains your IP address Netmask.

([0-9])
• Gateway
If PBXware is located in LAN, set the gateway IP address here.

(ex. 192.168.1.1)
([0-9])
• DNS server
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If PBXware is located in LAN, set the preferred DNS server IP address here
(ex.192.168.1.1)
([0-9])
• DHCP Server
If there is no DHCP server on LAN, PBXware can start its own and provide
UADs/Phones with this service. If you already have DHCP server on your
network, choose 'No'.
([0-9])
Server Details (Vserver version)
Server Details (Vserver version)
These are system and network fields necessary for proper system operation
• Root Password
PBXware prompts for this password during the system/ssh login and when
accessing system services through the interface.
(ex. do34ffjk)
([a-z] [0-9])
• Confirm Password
Re-type the Root Password entered in the field above.

(ex. do34ffjk)
([a-z] [0-9])
• Time zone
Select the the time zone, in which PBXware is located, ex. USA/East-coast.
(USA/East-coast)
(Select box)
• Hostname
The name given to the machine which will identify the system on the network,
ex. myhost.
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(ex. myhost)
([a-z] [0-9])
• Server name
Virtually the same as the 'Hostname' field, only this name will appear during
system notifications, sent emails, etc.
(domain.com)
([a-z] [0-9])
Licensing
Licensing allows you to enter a valid non-active license and displays the MAC address
with which the license will be valid. If the system has more than one network adapter,
all found MACs will be shown for selection.
Licensing
• License Type
Select a system license type. The system comes with two license types: FREE
license and 'Requested License'
(Select box)
• License Number
Enter the PBXware license number, select the MAC address if more than one is
present and click 'Next' to register the PBXware
([a-z] [0-9])
• MAC
The MAC address associated with the PBXware.

NOTE: The system must have access to a fully operational Internet connection
in order to license the system. If more than one MAC address is present, select
the one you wish to associate with PBXware and click 'Next'.
(Select box)
Locality
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Locality
Locality allows for 'local' system values to be entered in order to setup all the necessary
values for normal system operation.
• Country
Select the country in which PBXware is located, ex. United Kingdom.

(Select box)
• Indications
Typical telephony sounds PBXware will use. A different signal is heard when
the handset is picked up in different countries. This field will be set
automatically. If your country is not on the list, select 'Other'.
(Select box)
• Area Code
The area code of the city where PBXware is located, for example, if your
PBXware is in New York, you would set '212' here.
([0-9])
• Number Of Digits
The number of digits PBXware will associate with local extensions. If this field
is set to '4', all local extensions will have a range from 1000-9999.
(0-9)
WARNING: After you create first extension, queue or an agent on your system, you
will not be able to change this setting anymore.
• Police/Fire/Ambulance
The number of Emergency Services in the area where PBXware is located.

PBXware has an option to dial these emergency services through certain trunks.
If all trunk channels are busy, an active call will get dropped in order to dial
these numbers.
(If in the USA, for example, set these fields to '911')
(Select box)
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Music On Hold
The system comes with a default 'silence' music on hold sound file, or royalty free
music on hold file which can be changed later on.
Music on Hold
TIP:
This field is available for: Virtual Server, VMWare, CD, and Appliance PBXware
packages
• Royalty Free Music On Hold
This will upload royalty free music which can be used as default music on hold
but it can be changed later on.
(Option button)
Trunk
The trunks step will try to detect supported trunk devices present on the system. Once
detected, the wizard will automatically create a trunk based on the most common
configuration values.
Trunk
• Detected devices
If the setup wizard detects any hardware devices (cards), they will be listed here.
(Display)
• Default Destination
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If PBXware has no DIDs set, all incoming calls will go to this destination. Set
'1000', for example, to make all calls coming from trunks go to this extension.
([0-9])
Confirm
Finally, the confirmation step allows for all values to be revised and to either finish the
wizard or start all over from the beginning. If 'Confirm and finalize' is clicked, the setup
wizard will finish and the browser will be redirected to the system login screen.
Trunk
• Repeat the wizard
If you click this button all of the provided details will be reset and started again
from step one.
(Command button)
• Confirm and finalize
Click this button to confirm the data provided in the setup wizard and finalize
the installation. All provided details will be applied and the setup wizard will
redirect you to the PBXware login screen.
(Command button)
Logging into the system
Login screen
In order to login to the system, please point your browser to: http://$IPADDRESS/ (For
Example:http://192.168.1.2/)
• E-mail
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This is administrators e-mail address, address you have set in the initial setup
wizard.
([a-z] [0-9] [@_.-])
• Password/PIN
This is administrators password, password you have set in the initial setup
wizard.
([a-z] [0-9])
Updates
In order to update PBXware to the latest version follow these steps:
1. Login to the PBXware web interface

2. Navigate to 'Site Settings: Updates' and click on the 'Updates' button
3. Click on 'Start' and wait until the system shows the interface again
License Updates
To upgrade your PBXware license, do as follows:
1. Login to the PBXware web interface

2. Navigate to 'Site Settings: Updates'
3. Enter the system root username and password
4. Click on 'License' button
5. Enter the license number in the 'License Number' field (e.g. 0A9DS8F7)
6. Click on the 'Save' button
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MT 3.8.5 System Overview
Administration Interface
The administration interface consist of applications, administration and actions menus
with all data displayed in data area.
• Applications Menu
The applications menu provides access to all applications, site settings, SM

settings, log out, and help links.
PBXware
Clicking on PBXware will display its menu showing all available sections.
Site Settings
This link allows managing of system users, groups, backup, sessions, updates
and licensing.
SM Settings
Options of system administrators management and currently available sites can
be found here.
Help
A link to help documentation.
Logout
This link allows user to log out.
Select language
Drop down menu for PBXware GUI language selection.
• PBXware menu
The PBXware menu is located on the left and displays all administration
sections available.
• Data Area
Data Area displays all results from various actions performed.
• System Actions Menu
The system actions status on the top right allows viewing of the system status
and performing of start, stop, and restart actions on one or more servers.
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IMPORTANT:
Depending on what you have selected in the 'Select a tenant:' menu you will have a
different choice of options in the menu below.
If you are on the master tenant, you will have the ability to create/edit systems trunks
and DIDs as part of master tenants control features.
If one of the slave tenants is selected, you will control its PBX features which is part of
slave tenants functionality as PBXes.
Role Based Administration
Site users
System administration can be delegated to various users in order to perform role based
administration. An unlimited number of users and groups can be created by system
administrator(s). Each user will only have access to the sections of PBXware MT menus
according to group membership permissions.
This is commonly used to allow management, operators, supervisors, etc access to the
sections of the PBXware MT to which they have adequate knowledge and experience.
Standard and Advanced Options
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Advanced Options
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PBXware MT has been designed with simplicity and extensive configuration options as
primary goals. In order to achieve both goals, administration has standard and advanced
modes throughout all sections.
Standard Options
Standard mode is designed to allow an easy 'no brainer' method of administration as the
level of configuration knowledge needed is based on common fields of information
(Name, email address, etc.). This is possible to achieve since PBXware MT uses a
powerful template system which pre-configures all advanced options leaving only
common information values to be entered.
Advanced Options
Advanced mode options, on the other hand, require much better system knowledge in
order to fine tune the system settings for various applications and usages. This manual
tries to provide detailed information for advanced options. However, administrators
should bear in mind that extensive training and hands on experience is required in order
to be able to administer advanced sections effectively. For simplicity, all options
available in the standard options are not repeated in this manual under the advanced
options.
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MT 3.8.5 Extensions
Extensions are associated with all UADs/Phones registered to the current slave tenant.
On Multi Tenant PBXware, this menu item is only available when you are editing a
tenant, since the master tenant is used for controlling the system behavior and tenants
functionality.
Extensions
Contents
• 1 System
o 1.1 Search
o 1.2 Add/Edit Extension
o 1.3 Adding Multi Extensions
o 1.4 Advanced Options
o 1.5 Permissions
• 2 Ring Groups
o 2.1 Add/Edit Ring Group
• 3 Departments
o 3.1 Add/Edit Department
System
System Extensions lists all local and remote UADs/Phones connected to the current
tenant with the following details:
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System
• Name
Full name of the user to which the device is registered

(ex. Peter Doyle)
(Display)
• Extension
UAD/Phone extension number

(ex. 1111)
(Display)
• User Agent
UAD/Phone type
(ex. Yealink T38P)
(Display)
• Status
UAD/Phone system status

(ex. Active/Inactive)
(Display)
• Procotol
Protocol used by the UAD/Phone

(ex. SIP/IAX)
(Display)
• Edit
Edit UAD/Phone configuration

(Button)
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• Delete
Delete UAD/Phone from the system

(Button)
Search
The search bar filters extensions by name, e-mail, and number.
Search form
• Search
Provide a search phrase here and hit enter to filter the records.
([a-z][0-9])
• Name
The search filter should be applied to the names UADs are registered to.
Check the box to search the names.
(Check box)
• E-mail
The search filter should be applied to email addresses associated with the UADs.
Check the box to search the email addresses.
(Check box)
• Number
The search filter should be applied to extension numbers.

Check the box to search extension numbers.
(Check box)
• MAC
The search filter should be applied to MAC numbers of UADs.

Check the box to search for MAC numbers
(Check box)
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Add/Edit Extension
The procedure for adding a new system extension is divided into two steps. In the first
step, the UAD/Phone type and extension location need to be provided. In second step,
basic UAD/Phone information such as user's name and email address is provided.
Add/Edit Extension
TIP:
By default, a 'Single Extension' will be created. 'Advanced Options' offer the facility to
add multiple extensions as well. For more information, check the 'Adding Multi
Extensions' chapter.
• UAD (User Agent Device)
Select the model of the new system UAD/Phone.

If the UAD/Phone is not listed here, navigate to 'Settings: UAD' Edit the desired
UAD/Phone and set its 'Status' to 'Active'. Now, the UAD/Phone will be available in this
list.
(ex. Linksys SPA-941)
(Select box)
• Location
Select the location of the new UAD/Phone. Location refers to whether the UAD/Phone
is in 'Local' or 'Remote' network.
(ex. Local/Remote)
(Select box)
In the second step, basic UAD/Phone information is set.
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Add/Edit Extension
TIP:
Since this is an extension on a tenant you will see that the Username is prefixed with a
tenant code, which is required for a UAD/Phone to register to the system. Nevertheless,
when you register you will be able to dial other users on the tenant with only their
extension number.
• Extension
System extension number

By default, this field is automatically populated, but can be changed to any Extension
number.
(ex. Setting '1008' here will create a new system extension with the same network
number. By default, this field is automatically populated, but can be changed to any
Extension number).
([0-9])
• Name
Full name of the person using the Extension. This name is sent in a Caller ID
information For example, setting name 'Joanna Cox' in this field will display the name
on the other UAD/Phone display when the call is made.
([a-z][0-9])
• E-mail
Email address associated with the extension and used for various system notifications
(ex. Setting '[email protected]' here will transfer all Voicemail notifications,
Extension PIN and other details to this email)
([a-z][0-9])
• Department
Department to which extension will belong to. This is used so the Bicom Systems
gloCOM can group extensions depending on which department they belong to.
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(ex. Select "Sales" and when sorted in gloCOM, this extension will be shown in Sales
department group).
(Select box)
• Billing:
Turn Billing on or off for the current extension

(ex. Yes, No, N/A)
(Option buttons)
• Service Plan
Service plan applied to the extension

(ex. Select among available service plans to apply its rates to the extension)
(Select box)
• Slave
Set whether two extensions should share the same billing funds. If ext 1000 has 100.00
in credit, enable this option and set 'Slave Account Code'='1001'. Now, any call made
by these two extensions will take the credit off the 1000 extension.
(Option buttons)
• Username
Username used by the UAD/Phone for the registration with the PBXware MT
By default, this field is the Extension network number prefixed with tenant code and
cannot be changed.
(ex. In this case this value is set to '30010008').
([0-9])
• Secret
Secret/Password used by the UAD/Phone for registration with the PBXware MT

By default, this field is automatically populated but can be changed to any value
(ex. t8C1OGvK)
([a-z] [0-9])
• PIN (Personal Identification Number)
Four digit number used for account authorization.

NOTE: This number must always be four (4) digits long
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(ex. If the PIN for this extension is set to '8474', provide it when asked for it by the
PBXware MT when checking your Voice inbox or other Enhanced Services)
([0-9])
TIP:
• After the extension is created, the 'Permissions' group will be editable for the
administration.
• Do not paste a value to the 'Name' and 'Email' fields, but please type it in. If these
values are pasted, 'Advanced options' will need to be opened and the system will
prompt for missing values.
• Once the extension is created, the 'Save & Email' button becomes available. This
command sends Extension details on the provided 'E-mail' address.
Adding Multi Extensions
Add multiple extensions
There are two ways to add multiple extensions to PBXware MT:
• Manually
To create a list, manually provide details to 'Name', 'Email', 'Ext', 'Secret', 'PIN' [,
'MAC'] fields and click on the Add(+) icon.
• Uploading a '.csv' file
To Upload a '.csv' file:
• Open a text editor on your desktop

• Add the following lines, for example ('Name', 'Email', 'Exit', 'Secret', 'PIN', [,
'MAC'])
John Doe,[email protected],4444,1234,4444
Joanna Cox,[email protected],5555,2345,5555
• Save file as 'ext.csv'

• Click on the 'Browse' button
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• Select 'ext.csv' from the Desktop
• Click on the 'Upload' button
• New Extensions will be created
• Single/Multiple Extension
Switch between the single and multi extension adding options

(Option button)
Advanced Options
Advanced Options
Klicking Advanced Options button will show advanced configuration options and fields
that were hidden previously.
• UAD
UAD (User Agent Device) are various IP phones, soft phones, ATA (Analog Telephone
Adaptors), and IAD (Integrated Access Devices) used for system extensions. PBXware
supports a wide range of UAD using SIP, IAX, MGCP, and ZAPTEL protocols. In case
your phone is in supported UAD list, and UAD is enabled on your PBXware, you will be
able to choose UAD that match the phone registering to the extension.
• Location
This option is related to Auto Provisioning function of PBXware, extensions located in

same LAN as PBXware have to be set to Local while extensions connecting to PBXware
from WAN should be set to Remote.
General
The following options are used frequently and are mostly required for normal extension
operation. Some of these fields are pre-configured with the default values. It is not
recommended to change these unless prompted to do so while saving the changes.
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General Fields
• Extension
Extension number. Number you dial on in order to reach local PBXware user
associated with this extension.
• Title
Users title like Mrs, Mr, and such

(ex. Mrs)
([a-z])
• Name
Name of user associated with extension e.g. John Lebowski.
• E-mail
E-mail address associated with user of the extension. For example, this e-mail address
can be used to send out extensions account details to the user.
• Location
User's location
(ex. Location like Street, City, or State)
([a-z][0-9])
• Department
Department in company with which user is associated with e.g. Development.
• User Type
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Extensions can be set to make calls only, receive calls only or both make and receive
calls
(ex. Select 'User' to make the calls only; 'Peer' to receive the calls only; or 'Friend' for
both, to make and receive calls)
(Select box)
• DTMF Mode (Dual Tone Multi-Frequency)
A specific frequency, consisting of two separate tones. Each key has a specific tone
assign to it so it can be easily identified by a microprocessor.
This is a sound heard when dialing digits on touch-tone phones. Each phone has
different 'DTMF Mode'.
(ex. By default, this field is populated automatically for supported devices. If adding
other UAD/Phone select between 'inband', , 'rfc2833' or 'info' options)
(Select box)
• RFC2833 Compensate (1.2):
Compensate for pre-1.4 DTMF transmission from another Asterisk machine. You must
enable this option or DTMF reception will not work.
(ex. Yes, No, N/A)
(Option Buttons)
• Context
Every system extension belongs to a certain system context. Context may be described
as a collection/group of extensions. Default context used by the PBXware MT per
tenant is 't-XXX' (where XXX is tenant number) and cannot be changed.
• Status
Extension status/presence on the network.

Rather than deleting the extension and then recreating it again later on, the extension
can be activated/deactivated using this field.
(ex. Setting this field to 'Not Active' will disable all calls to this extension).
Options:
Active - Extension is active, it can make and receive calls.
Not Active - Extension is not active and it can't make nor receive calls.
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Suspended - Extension is suspended and can't make calls to numbers other than those
defined as Emergency Service numbers in Settings -> Servers -> Edit Server -> Locality
(section) -> Emergency Services,
(Select box)
• Show In Directory:
Whether the extension should be shown in the directory or not

(ex. Yes, No, N/A)
(Option buttons)
Authentication
These options are used for UAD/Phone authentication with PBXware MT
Authentication
• Username
Username used by the UAD/Phone for the registration with PBXware MT

(ex. By default, this field is the same as the extension network number and cannot be
changed. In this case, this value is set to '1008').
([0-9])
• Authname
Name used for authentication with the sip provider

Example:
If you set this field to 12345, for example, the sent SIP header will look like
[email protected], for example
([0-9])
• Auth
Auth is the optional authorization user for the SIP server

(ex. 44000)
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([a-z][0-9])
Strong Password Generator
• Secret
Secret/Password used by the UAD/Phone for registration with PBXware MT

(ex. By default, this field is automatically populated, but can be changed to any value)
([a-z][0-9])
NOTE: In PBXware 3.8.2 we introduced strong password enforcement, which means

that secret must meet certain criteria in order to be accepted otherwise, PBXware will
display error message stating that secret is too weak.
Strong Password Requirements
Secret has to meet the following criteria in order to be accepted:
• It must be at least 8 characters long
• It must contain at least 1 uppercase
• It must contain at least 1 lowercase
• It must contain at least 1 digit
• It must contain at least 1 special character ()
• Allowed characters are: a-z, A-Z, 0-9, ! % * _ -
In order to make it easier for our users, we also implemented password generator, that
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will automatically generate strong password that meets above criteria with a single
mouse click on a key icon located on a side of Secret field.
• PIN (Personal Identification Number)
Four digit number used for account authorization.

NOTE: This number must always be four (4) digits long
(ex. If the PIN for this extension is set to '8474', provide it when asked for it by
PBXware MT when checking your Voice inbox or other 'Enhanced Services')
(0-9)
IAX Extensions only
• Auth Method
Authorization method used for IAX extensions, can be set to:
• none
• plaintext
• md5
• rc4
• rsa
• RSA Key
RSA key used for authorization, if preferred auth method is 'rsa' then RSA key needs to
supplied to this field.
([0-9][a-z][A-Z])
• Encryption
Whether to enable encryption of IAX data stream. For this to work, you must choose
'md5' auth method, for example Yes, AES-128.
Billing
These options are used for billing of incoming and outgoing calls. The extension is
assigned to a service plan and its call rates and additional billing options are set here as
well.
32
Billing settings
• Billing
This option will enable/disable billing on extension.
• Reset Inclusive
Reset extension inclusive minutes, click on this button and confirm with 'Yes' to reset
inclusive minutes.
(Button)
• Credit/Debit
Opens a window for adding extension credit/debit.

(Button)
Credit/Debit
Credit/Debit window
• Type
Billing type, select whether billing is credit or debit.

(Select box)
• Amount
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Billing amount, if the billing type is in Euros, and you add 100 here, 100 Euros will be
added to the extension amount.
([0-9])
• Ref No
Billing reference number, depending on how your company bills clients, the invoice
number can be assigned here, for example.
([a-z][0-9])
• Notes
Additional billing notes.

([a-z][0-9])
• Send
This will finalize billing action, fill in all previous fields and click this button to add
funds.
(Button)
Once funds are added, the following details will be displayed:
• Date: Time and date of the payment

• User: The username used for login to the system of the user who added the
funds
• Ref No: Billing reference number
• Notes: Additional billing notes
• Amount: Amount of funds added
• Type: Billing type
NOTE: Buttons Reset Inclusive and Credit/Debit will not be displayed unless Billing is
enabled.
• Slave
Set whether two extensions should share the same billing funds. If ext 1000 has 100.00
in credit, enable this option and set 'Slave Account Code'='1001'. Now, any call made
by these two extensions will take the credit off the 1000 extension.
(Option buttons)
• Master Account Code
Set the master account code (extension number) from which the current extension is
using funds. If ext 1000 has 100.00 in credit, enable this option and set 'Slave'='Yes'
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and set this field to '1001'. Now, any call made by these two extensions will take the
credit off the 1000 extension.
([0-9])
• Reminder Balance
Account balance at which a reminder should be sent to the user. If this field is set to
10, the user will receive an email notification when the account balance reaches this
amount.
([0-9])
• Credit Limit
The maximum amount that the system will extend to the billing account. If this field is
set to '10' and the account balance has dropped down to '0', your account will still
have '10' units in available funds.
([0-9])
• Service Plan Date (dd-mm-YYYY)
When the current Service Plan was set so inclusive minutes can be reset. If this field is
set to 12-06-2008, inclusive minutes will be reset on 12th day of each month. So, if all
5 inclusive minutes were used by this day, inclusive minutes will be reset back to 5
minutes.
([dd-mm-YYY])
• Enable Limits
This option set the limits on the current extension to Yes, No, N/A.
(Option buttons)
• Limit Type
This option set limits to be applied Daily or Monthly.

(Select box)
• Soft Limit
Depending on Limit Type, when the extension reaches Soft Limit, it will email the
person in charge of billing. Set 10 here if you want an email sent when the user hits
that amount when calling.
([0-9])
• Hard Limit
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Depending on the Limit Type, when an extension reaches Hard Limit, the system will
block this extension from making any further calls. Set 20 here if you want the system
to block this extension from making any calls.
([0-9])
• Notification Email
What email should be used when the user reaches its Soft Limit.
([email protected])
([a-z][0-9])
Billing Info
This information displays the extension's billing information: amount left, inclusive
minutes left, etc.
• Account Balance
Displays the available account balance - the exact sum spent by the user.
(ex.If the user has 100 units of credit, 100 units + the credit limit can be spent. If this
amount displays a negative value (e.g. -4.00000) that means that the account balance
has reached 0 and the credit limit is being used).
(Display)
• Available Funds
Displays available account funds (account balance + credit limit).

(ex. If the user account balance has 100 units + 10 credit limit units, 110 units will be
displayed here).
(Display)
• Inclusive Minutes Left
Displays the inclusive minutes left. As long as there is any inclusive time left, billing is
not calculated for outgoing calls.
(ex. You'll see the inclusive minutes left in the following form '0d 0h 4m 25s').
(Display)
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• Creation Date
Extension creation date.

NOTE: If your system was updated to newer version, old extensions will have this field
displaying 'unknown' and all new extensions will display extension creation date.
(ex. 14-06-2007 12:30:36)
(Display)
• First Use Date
Date/Time of the first extension use.

(ex. 11 Jun 2007 18:58:25)
(Display)
• Last Use Date
Date/Time of the last extension use.

(ex. 11 Jun 2007 19:25:12)
(Display)
Permissions
Permissions settings
Destinations
These options grant/deny certain local/worldwide destinations, conferences, enhanced

services, or call monitoring to your edited extension. If the image below is displayed, all
destinations are allowed for the user extension. Should extension permissions be
changed, click the 'Set destinations manually' button.
37
Destinations settings
Manually, destinations are set through the following groups:
• Remote - E164 PSTN destinations, ITSPs, other VoIP networks etc.

• Local - All destinations within the system/network (Extensions, IVR, Queues,
Conferences...).
• Other Networks - Other PBX networks we are connected to.
Authorized
PIN Required
Not Authorized
Enhanced Services
Enhanced Services allows users to fully adjust settings like Caller ID, Call Pickup, Call
Filters & Blocking, Call Forwarding etc.
For detailed information on Enhanced Services click the link below:
Enhanced Services
38
Network Related
Network related settings
These options set important network related values regarding NAT, monitoring and
security.
• Transport:
Type of transfer protocol that will be used on PBXware.

UDP (User Datagram Protocol) - With UDP, computer applications can send messages,
in this case referred to as datagrams, to other hosts on an Internet Protocol (IP)
network without prior communications to set up special transmission channels or data
paths.
TCP (Transmission Control Protocol) - provides reliable, ordered, error-checked

delivery of a stream of octets between programs running on computers connected to
an intranet or the public Internet.
TLS (Transport Layer Security) - cryptographic protocol that provide communication

security over the Internet.[1] They use asymmetric cryptography for authentication of
key exchange, symmetric encryption for confidentiality, and message authentication
codes for message integrity.
Type: Checkbox
• Encryption:
This option enables or disables encryption in PBXware transport.

Options: Yes, No, N/A.
• NAT (Network Address Translation)
Set the appropriate Extension - PBXware NAT relation.

If extension 1000 is trying to register with the PBXware from a remote
location/network and that network is behind NAT, select the appropriate NAT settings
here:
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• yes - Always ignore info and assume NAT
• no - Use NAT mode only according to RFC3581
• Default (rport) - this setting forces RFC3581 behavior and disables symmetric
RTP support.
• Comedia RTP - enables RFC3581 behavior if the remote side requests it and
enables symmetric RTP support.
(Option buttons)
• Direct Media
This option tells the Asterisk server to never issue a reinvite to the client, if it is set to
No. Select Yes if you want Asterisk to send reinvite to the client.
(Option button)
• Direct RTP setup:
Here you can enable or disable the new experimental direct RTP setup. Setting this
value to yes sets up the call directly with media peer-2-peer without re-invites. Will
not work for video and cases where the callee sends RTP payloads and fmtp headers in
the 200 OK that does not match the callers INVITE. This will also fail if directmedia is
enabled when the device is actually behind NAT.
Options: Yes, No, N/A
• Qualify
Timing interval in milliseconds at which a 'ping' is sent to the UAD/Phone or trunk, in

order to find out its status(online/offline). Set this option to '2500' to send a ping
signal every 2.5 seconds to the UAD/Phone or trunk. Navigate to 'Monitor: Extensions'
or 'Monitor: Trunks' and check the 'Status' field.
([0-9])
• Host
Set the way the UAD/Phone registers to PBXware. Set this field to 'dynamic' to register
the UAD/Phone from any IP address. Alternately, the IP address or hostname can be
provided as well.
([dynamic][a-z][0-9])
• Default IP
Default UAD/Phone IP address. Even when the 'Host' is set to 'dynamic', this field may
be set. This IP address will be used when dynamic registration could not be performed
or when it times out.
• NOTE: UAD/Phone must be on static IP address.
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([0-9])
• Use RTP source address for T.38 packets (1.2)
Use the source IP address of RTP as the destination IP address for UDPTL packets if the
nat option is enabled. If a single RTP packet is received Asterisk will know the external
IP address of the remote device. If port forwarding is done at the client side then
UDPTL will flow to the remote device.
(ex. Yes, No, N/A)
(Option Buttons)
Caller ID
The caller's name and number displayed here are sent to the party you call and are
shown on their UAD/Phone display. The information you see here is taken from the
extension number and user name. To set different Caller ID information, please go to
'Enhanced services: Caller ID' and set new information there.
Caller ID
• Set Caller ID
Enable 'Caller ID' service

(ex. Set this option to 'Yes' to enable the Caller ID service)
(Option buttons)
• Caller ID
Extension Number and Name that are displayed on dialed party UAD/Phone display
(ex. These options are read-only. Caller ID information can be changed only through
'Enhanced Services')
(Read-only)
• Caller ID Presentation
The way Caller ID is sent by the Extension
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If PBXware MT is connected to a third-party software and there are problems with
passing the Caller ID information to it, applying different 'Caller ID Presentation'
methods should sort out the problem
(ex. Presentation Allowed, Not Screened)
(Select box)
• Hide CallerID for Anonymous calls
When you set this option to Yes, incoming calls with Anonymous as number but have
CallerID set, are then formatted as Anonymous <anonymous>.
(ex. Yes, No, N/A)
(Option buttons)
• Ringtone for Local calls
If you know which phone is registered on this extension, you can set a custom ring
tone for local calls.
(ex. If your phone is SPA941 you could set <Simple-2>)
([a-z][0-9])
• Only Allow Trunk CallerID within DID range
When you assign an extension to a customer and assign some DIDs to it, customer can
make calls through that extension with CallerIDs that match its DID numbers. If a
customer tries to make a call with a CallerID that doesn't match any of the DIDs
assigned to him, the call will not be allowed.
(ex. Yes, No, N/A)
(Option buttons)
Call Properties
These options fine-tune incoming/outgoing call settings.
Call Properties settings
42
• Ringtime
UAD/Phone ring time.
Example:
Time in seconds that the UAD/Phone will ring before the call is considered unanswered
(default: 32).
([0-9])
• Incoming Dial Options
Advanced dial options for all incoming calls.

Example:
Please see below for a detailed list of all available dial options (default: tr).
([a-z])
• Outgoing Dial Options
Advanced dial options for all outgoing calls.

Example:
Please see below for a detailed list of all available dial options (default: empty).
(a-z)
• VoiceMaster PIN
This is a PIN number that is issued along with a dial string to the VoiceMaster system.
(ex. 1234)
([0-9])
Dial Options:
• t - Allow the called user to transfer the call by hitting #

• T - Allow the calling user to transfer the call by hitting #
• r - Generate a ringing tone for the calling party, passing no audio from the called
channel(s) until one answers. Use with care and don't insert this by default into all of
your dial statements as you are killing call progress information for the user. Really,
you almost certainly do not want to use this. Asterisk will generate ring tones
automatically where it is appropriate to do so. 'r' makes it go the next step and
additionally generate ring tones where it is probably not appropriate to do so.
• R - Indicate ringing to the calling party when the called party indicates ringing, pass no
audio until answered. This is available only if you are using kapejod's bristuff.
43
• m - Provide Music on Hold to the calling party until the called channel answers. This is
mutually exclusive with option 'r', obviously. Use m(class) to specify a class for the
music on hold.
• o - Restore the Asterisk v1.0 Caller ID behavior (send the original caller's ID) in Asterisk
v1.2 (default: send this extension's number)
• j - Asterisk 1.2 and later: Jump to priority n+101 if all of the requested channels were
busy (just like behaviour in Asterisk 1.0.x)
• M (x) - Executes the macro (x) upon connect of the call (i.e. when the called party
answers)
• h - Allow the called party to hang up by dialing *
• H - Allow the caller to hang up by dialing *
• C - Reset the CDR (Call Detail Record) for this call. This is like using the NoCDR
command
• P (x) - Use the Privacy Manager, using x as the database (x is optional)
• g - When the called party hangs up, exit to execute more commands in the current
context.
• G (contextêxten^pri) - If the call is answered, transfer both parties to the specified
priority; however it seems the calling party is transferred to priority x, and the called
party to priority x+1
• A (x) - Play an announcement (x.gsm) to the called party.
• S (n) - Hang up the call n seconds AFTER the called party picks up.
• d: - This flag trumps the 'H' flag and intercepts any dtmf while waiting for the call to be
answered and returns that value on the spot. This allows you to dial a 1-digit exit
extension while waiting for the call to be answered - see also RetryDial
• D (digits) - After the called party answers, send digits as a DTMF stream, then connect
the call to the originating channel.
• L (x[:y][:z]) - Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms)
Only 'x' is required, 'y' and 'z' are optional. The following special variables are optional
for limit calls: (pasted from app_dial.c)
o + LIMIT_PLAYAUDIO_CALLER - yes|no (default yes) - Play sounds to the caller.
o + LIMIT_PLAYAUDIO_CALLEE - yes|no - Play sounds to the called party.
o + LIMIT_TIMEOUT_FILE - File to play when time is up.
o + LIMIT_CONNECT_FILE - File to play when the call begins.
o + LIMIT_WARNING_FILE - File to play as a warning if 'y' is defined. If
LIMIT_WARNING_FILE is not defined, then the default behavior is to announce
('You have [XX minutes] YY seconds').
• f - forces callerid to be set as the extension of the line making/redirecting the outgoing
call. For example, some PSTNs don't allow Caller IDs from other extensions than the
ones that are assigned to you.
• w - Allow the called user to start recording after pressing *1 or what defined in
features.conf, requires Set(DYNAMIC_FEATURES=automon)
• W - Allow the calling user to start recording after pressing *1 or what defined in
Groups
These options define who is allowed to pickup our calls, and whose calls we are
allowed to pickup.
44
Groups settings
• Call Group
Set the Call Group that the extension belongs to. Similar to 'Context' grouping, only
this option sets to which call group the extension belongs.
(ex. 3)
(Select box)
• Pickup Group
Set which groups the extension is allowed to pickup by dialing '*8'.

(ex. 4)
(Select box)
TIP: Grouping works only within a technology (SIP to SIP or IAX to IAX)
Example:
Extension A:
• Call Group = 1
• Pickup Group = 3,4
Extension B:
• Call Group = 2
• Pickup Group = 1
• If A is ringing, B can pickup the ringing call by dialing '*8'.
• If B is ringing, A cannot pickup the ringing call because B's Call Group = 2, and A
can pickup only Call Groups 3 and 4.
NOTE: To be able to select Call Group and Pickup Group they have to be assigned to
the tenant in Settings -> Tenants -> edit tenant -> Numbering Defaults (section) -> Call
groups/Pickup groups.
45
Trunks
These options enable extensions to use custom default trunks for all outgoing calls.
Trunks
• Primary/Secondary/Tertiary Trunk:
Set the default trunks for all routes dialed from this extension.
If the connection is not established through the primary, the secondary trunk is used,
etc. Default trunks can be set per extension and on the Settings->Default Trunk, on a
Slave. Please look at the 'Precedence' section.
(Select box)
• Override System LCR
This option tells the systems that when making calls, they should omit checking LCR.
(Option buttons)
Call Control
These options set the number of simultaneous incoming and outgoing extension calls.
Call Control
• Incoming Limit:
Sets the maximum number of simultaneous incoming calls. If an extension receives

more incoming calls than set here, they are all redirected to the extension voice-box
(ex. 2)
([0-9])
46
• Outgoing Limit:
Sets the maximum number of simultaneous outgoing calls. The outgoing call can be
placed on hold and another call can be made from the same extension. However, this
feature has to be supported by the UAD/Phone
(ex. 2)
([0-9])
• Play sound on exceeded limit:
If you try to make more calls than allowed in the Outgoing Limit, you will be played a
message that the limit has been exceeded.
(ex. Yes, No, N/A)
(Option button)
• Send e-mail on exceeded limit:
Whether or not to send a notification mail on the exceeded limit.

(ex. Yes, No, N/A)
(Option buttons)
• Notification e-mail:
E-mail address to which notification mail should be sent if the number of calls exceed
the limit.
(ex. [email protected])
([a-z][0-9] @)
IAX Extensions only
• Notransfer
Prohibit Asterisk from stepping out of the media path and connecting the two
endpoints directly to each other. This, of course, affects your CDR and billing
information
(ex. Yes, No, N/A)
47
(Option buttons)
• Send ANI
Whether to send ANI along with CallerID

(ex. Yes, No, N/A)
(Option buttons)
• Trunk
Whether to use IAX trunking. IAX Trunking needs support of a hardware timer
(ex. Yes, No, N/A)
(Option buttons)
Voicemail
These options mimic the functions of an answering machine but with many additional
features added. Voice messages are saved on central file-system location instead on a
UAD/Phone.
• Accessing voice-box:
To access a voice-box, dial '*123', enter the extension PIN, and follow the instructions.
• Leaving a voice message:
When the user is transferred to voice-box, 'Please leave your message after the tone.
When done, hangup or press the # key' message will be heard. Two options are
available:
1. Leave a voice message (ended by pressing '#' key or by hanging up), or

2. Reach an operator by dialing '0'
If '0' is dialed, the 'Press 1 to accept this recording, otherwise please continue to hold'
message will be heard. Two options are available:
1. Press '1' to save your message, after which the operator will be dialed. The 'Please
hold while I try that extension' message will be heard, or
2. Continue to hold, which will delete any left messages, after which the operator will be
dialed. 'Message deleted, please hold while I try that extension' message will be heard.
• File - system usage:
With continuous tone for 60 seconds:
• wav49 = 91.0kb
• wav = 863.0kb
48
• gsm = 91.0kb
With continuous silent tone for 60 seconds:
• wav49 = 0.38kb
• wav = 3.0kb
• gsm = 0.32kb
Voicemail
• Voicemail:
Enable the Voicemail service.

When the call is placed and no one picks up the handset after some time, the calling
party will be transferred to the dialed extension voice box and offered to leave a voice
message
(ex. Yes, No, N/A)
(Option buttons)
• Greeting-Mode:
If your Voicemail is turned on, you can set this option to yes to play a greeting and
then a busy sound
(ex. Yes, No, N/A)
(Option buttons)
• Mailbox
Mailbox extension number
49
(ex. This value is the same as the extension number and cannot be modified)
(Read-only)
• Name:
Full name of the user associated with the voice box.

(ex. This value is the same as the 'Name' field and cannot be modified).
([a-z][0-9])
• PIN: (Personal Identification Number)
Password used for accessing voicemail. The value of this field is set under
'Authentication: PIN'.
(ex. When B wants to access his voicemail, he is asked to authenticate with personal
4(four) digit PIN).
([0-9])
• E-mail:
E-mail address associated with the voice box. The value of this field is set under
'General: E-mail'.
(ex. When A calls B and leaves a voice message, B will get an email notification about
new voice message on this email address).
([a-z] [0-9] [@._-])
• Send e-mail
Whether or not to send an e-mail to the address given above

(ex. Yes, No, N/A)
(Option buttons)
• Pager e-mail:
Pager e-mail address associated with the voice box.

(ex. When A calls B and leaves a voice message, B will get a pager email notification
about a new voice message on this email address).
([a-z] [0-9] [@._-])
• Greeting message:
Greeting message played to users upon entering the voice box.
50
(ex. When A gets to B's voice box, the selected 'Greeting message' is played to A
before he is allowed to leave a message).
(Select box)
• Skip Instructions:
Skip the instructions on how to leave a voice message.

(ex. Once user A reaches the dialed voice box, if this option is set to 'Yes', A will hear
the 'Greeting message', and then be transferred directly to the 'beep' sound).
(Option buttons)
• Attach:
Send the voice message as an attachment to the user's email.

(ex. Once B gets the new voice message, if this option is set to 'Yes', the message
sound file will be attached to the new voicemail notification email).
(Option buttons)
• Delete After E-mailing:
Delete voice message after sending it as an attachment to the user's email.

(ex. Once B gets the new voice message, if this option is set to 'Yes', the message will
be deleted from the voice box after it has been emailed to B).
(Option buttons)
• Say Caller ID:
Announce the extension number from which the voice message has been recorded.
(ex. If this option is set to 'Yes', when checking voicemail, the 'From phone number
{$NUMBER}' message will be heard).
(Option buttons)
• Allow Review mode:
Allow B to review the voice message before committing it permanently to A's voice
box.
Example:
B leaves a message on A's voice box, but instead of hanging up, he presses '#'. Three
options are offered to B:
• Press 1 to accept this recording

• Press 2 to listen to it
51
• Press 3 to re-record your message
(Option buttons)
• Allow Operator:
Allow B to reach an operator from within the voice box.

Example:
B leaves a message on A's voice box, but instead of hanging up, B presses '#'.
'Press 0 to reach an operator' message played (Once '0' is pressed, user is offered the
following options):
• Press 1 to accept this recording (If selected, 'Your message has been saved.
Please hold while I try that extension' is played and operator is dialed)
• Or continue to hold (If B holds for a moment, 'Message deleted. Please hold
while I try that extension' is played and operator is dialed)
(Option buttons)
• Operatior Extension:
Local extension number that acts as an operator.

(ex. If A's voice box has an option 'Allow Operator' set to 'Yes', all users dialing '#0'
inside the voice box will reach this operator extension).
([0-9])
• Play Envelope message:
Announces the Date/Time and the Extension number from which the message was
recorded.
(ex. Once the voice box is checked for new messages, if this option is set to 'Yes',
'Received at {$DATE} from phone number {$NUMBER}' will be played, giving more
details about the message originator).
(Option buttons)
• Hide from directory:
This option will allow you to hide your extension from the Directory/BLF list.
(ex. Yes, No, N/A)
(Option buttons)
• Rings to answer
Number of rings before Voicemail answers the call

52
(ex. 5)
([0-9])
• Voicemail Delay:
How long to pause in seconds before asking the user for PIN/Password.
(ex. Some UADs/Phones have a tendency to garble the beginning of sound files.
Therefore, the user checking the voice box, when asked for a password, would hear
'...sword' instead of 'Password'. Setting this field to 1-2 seconds will provide a long
enough gap to fix this anomaly).
([0-9])
• Timezone:
Sets the correct date/time stamp.

NOTE: Timezones are taken from '/usr/share/zoneinfo' system directory
(ex. By setting the correct time zone, the user would always be notified of the exact
date/time voice message was left on their box. Set the correct time zone if the user is
located in a different time zone than PBXware MT).
(Select box)
Speakerphone Page Auto-Answer SIP Header
These options allow the caller to use a UAD in a public announcement system. If the
UAD fully supports this service, the call is accepted automatically and put on a
loudspeaker.
Speakerphone Page Auto-Answer SIP header
• Choose Device Type
Set predefined UAD/Phone type for this extension.

(ex. The header will be added automatically depending on the selected device).
(Select box)
• Custom Header
Set a custom UAD/Phone header for this extension.
53
(ex. If one of the predefined headers does not work, you might want to try setting a
custom header for this service. The custom header line to be used 'Call-Info:\;answer-
after=0').
([a-z][0-9])
Codecs
Codecs settings
Codecs are used to convert analog to digital voice signals and vice versa. These options
set preferred codecs used by the extension.
TIP:
If some of the desired codecs is disabled (cannot be selected), navigate to 'Settings:

Servers: Edit: Default Codecs' and enable them under the 'Local' group.
• Disallow
Set the codecs extension to not allowed to use.

(ex. This field is very unique. In order to work properly, this setting is automatically set
to 'Disallow All' and it cannot be modified).
(Ready-only)
• Allow
Set the codecs extension to allowed to use.

(ex. Only the codecs set under 'Settings: Server' will be available to choose from).
(Check box)
• Video Support
Set this option to Yes to enable SIP video support.

(Yes, No, N/A)
• Force codec on outbound trunk channel
54
With this option you can force codec use for outbound trunk calls.
(ex. iLBC)
(Select box)
• Auto-Framing (RTP Packetization)
If autoframing is turned on, the system will choose the packetization level based on
remote ends preferences.
(ex. If the remote end requires RTP packets to be of 30 ms, your PBXware system will
automatically send packets of this size if this option is turned on. Default is set to 20
ms and also depends on the codecs minimum frame size like G.729 which has 10 ms as
a minimum).
(Option buttons)
Codecs:
• ITU G.711 ulaw - 64 Kbps, sample-based, used in US

• ITU G.711 alaw - 64 Kbps, sample-based, used in Europe
• ITU G.723.1 - 5.3/6.3 Kbps, 30ms frame size
• ITU G.726 - 16/24/32/40 Kbps
• ITU G.729 - 8 Kbps, 10ms frame size
• GSM - 13 Kbps (full rate), 20ms frame size
• iLBC - 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
• Speex - 2.15 to 44.2 Kbps
• LPC10 - 2.5 Kbps
• H.261 Video - Used over ISDN lines with resolution of 352x288
• H.263 Video - Low-bit rate encoding solution for video conferencing
• H.263+ Video - Extension of H.263 that provides additional features that improve
compression over packet switched networks.
Recording
This group of options is used for the recording of all incoming/outgoing calls.
Recording settings
TIP:
• Laws in some countries may require notifying the parties that their call is being
recorded.
55
• Recorded calls, marked with icon, can be accessed from 'Self Care Interface' or
'Reports: CDR' PBXware' menu.
• Call are recorded in audio format set under 'Settings: Servers: Recordings Format'.
• Record Calls
Enable call recording service. Select 'Yes' to enable the service. All incoming/outgoing
calls will be recorded. If using call recording with many extensions, check server disk
space from time to time. Please see below for bit rates table.
(Options buttons)
• Silent
Set call recordings should be announced to the parties in a conversation. If Silent=No,

calling parties will hear a 'Recorded' or 'This call is recorded' message before their
conversation starts.
(Options buttons)
Disk Space Used By Call Recording
• wav49 = 84.5kb
• wav = 833.0kb
• gsm = 85.0kb
With continuous silent tone (without sound) for 60 seconds:
• wav49 = 84.0kb
• wav = 827.0kb
• gsm = 84.0kb
Auto Provisioning
These options enable PBXware MT to automatically provision the UAD/Phone.

Configuration files are downloaded from PBXware MT's TFTP server
Auto Provisioning
• Auto Provisioning:
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Enable auto provisioning service for this extension
(ex. Connect the UAD/Phone to PBXware MT without any hassle by providing
UAD/Phone MAC address (and optionally adding the Static UAD/Phone IP address and
network details))
(Option buttons)
• MAC Address (Media Access Control):
UAD/Phone MAC address

(ex. Provide the UAD/Phone address here. Its a 48-bit hexadecimal number (12
characters))
([a-z][0-9])
• DHCP (Dynamic Hosts Configuration Protocol):
Set whether the UAD/Phone is on DHCP or Static IP address

(ex. Set DHCP = Yes if UAD/Phone is on dynamic or DHCP = No if UAD/Phone is on
static IP address. If on static IP, you will have to provide more network details in the
fields below).
(Option buttons)
• Static IP:
Static UAD/Phone IP address

(ex. DHCP = No, has to be set. Provide the UAD/Phone static IP address here)
([0-9][.])
• Netmask:
UAD/Phone netmask
(ex. Netmask applied to UAD/Phone static IP address)
([0-9][.])
• Gateway:
Gateway IP address
(ex. Local area network gateway IP address)
([0-9][.])
• DNS Server1 and Server2 (Domain Name Server):
DNS Server IP address
57
(ex. Local area network DNS IP address (Usually the same as your gateway))
([0-9][.])
Presence
This option simply notifies you of whether device presence is enabled or disabled.
Supported UADs can be seen in the Settings->UAD menu.
Presence
• Presence Enabled:
Returns the information whether the phone is on call, ringing, or offline (not
registered).
(ex. Select 'Yes' to enable presence support, but all UADs/Phones don't support this
feature)
(Option buttons)
• Global Presence:
Enables presence like above option but when this option is turned on, it will enable
presence on all tenants on the system.
(ex. Yes, No)
(Option buttons)
Supported UADs:
• Snom 190(Firmware >= 3.60s), 320/360(Firmware >= 4.1)

• Polycom IP30x/IP50x/IP600
• Xten EyeBeam
• Grandstream GXP2000 (Firmware >= 1.0.1.13)
• Aastra 480i
• Aastra 9133i
CLI
show hints
-= Registered Asterisk Dial Plan Hints =-
1009 : SIP/1009 State:Idle
Watchers 0
Watchers 0
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1020 : SIP/1020 State:InUse
Watchers 0
1016 : SIP/1016 State:Unavailable
Watchers 0
Watchers 0
Watchers 0
1000 : SIP/1000 State:Ringing
Watchers 0
Watchers 0
Watchers 0
1234 : IAX2/1234 State:Unavailable
Watchers 0
7777 : IAX2/7777 State:Idle
Watchers 0
1017 : IAX2/1017 State:Unavailable
Watchers 0
----------------
- 12 hints registered
User Agent Auto Provisioning Template
User Agent Auto Provisioning Template
This option allows adding of additional settings to auto-provisioning template. Auto-

provisioning settings are generally defined in the 'Settings: UAD' and are custom set for
each device.
NOTE: Unless absolutely sure, do not change or add to this template.
Additional Config
Additional Config
This option is used for providing additional config parameters for SIP and IAX
configuration files. Values provided here will be written into these configuration files.
59
NOTE: Unless absolutely sure, do not change or add to this template.
Ring Groups
Ring Groups
Ring Groups are used to group a number of UADs/Phones into one network destination.
Each Ring Group is assigned a network number which, once dialed, rings all extensions
assigned to the group.
• Ring Group:
Ring group extension number

(ex. Accounts)
(Display)
• Extension:
Ring group extension number

(ex. Once a user dials this number, all destinations assigned to the ring group will ring
(e.g. 1111))
(Display)
• Destinations:
Extension Numbers assigned to a ring group

(ex. Once a ring group number is dialed, all destinations set here will ring at the same
time (e.g. 1001, 1002, 1003...))
(Display)
• Last Destination:
Last destination to be called if none of the destination extensions answer the call
(ex. 1010)
(Display)
• Edit
Edit the ring group configuration
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(ex. Click to edit the ring group configuration)
(Button)
• Delete
Delete a ring group from the system

Click to delete a ring group from the system
(Button)
Add/Edit Ring Group
Add/Edit Ring Group
Clicking on the 'Add Ring Group' or 'Edit' button will open the following ring group
options:
• Name:
Unique Ring group name

(ex.Set 'Accounts' here to create the same ring group)
([a-z][0-9])
• Extension:
Unique network number associated with the Ring group

(ex. When this number is dialed, all extensions associated with it will ring at the same
time)
([0-9])
• Extensions:
System extensions associated with the ring group

Example:
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Provide an extension list separated by commas here (e.g. 1001,1002,1003...). When a
ring group 'Extension' number is dialed, all extensions set here will ring at the same
time.
NOTE: If all destinations fail after 'timeout', 'Last Destination' will be called.
([0-9])
• Incoming Limit (per call):
If you have a scenario where call is sent from the current ring group to the second one
and the second one returns the same call back to first group, it will allow only this
much loops.
Example:
If this is set to 1 as it is by default, and the current ring group sends the call to the next
group (or any other object on the system), returning the same call from that object will
not be permitted as same call can enter this group only once.
NOTICE: system wide limitation for these 'loops' is 10.
Advanced Options
These options fine-tune ring group settings with additional options
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Advanced Options
• Greeting:
Greeting sound file played to callers when the Ring group is dialed
Example:
Select 'greeting-default-attendant', for example. Any user that calls this ring group will
hear this sound file played to them before all ring group extensions are dialed
(Select box)
• Answer on undefined greeting:
If this option is turned on, the ring group will not answer until the proper greeting is
selected.
(ex. Yes, No, N/A)
(Option button)
• Timeout Message
Sound file played to caller if his call does not get answered by any of the ring group
extensions.
NOTE: Sound file must have 'announce-' name prefix (e.g. 'announce-unavailable')
(ex. John dials ring group 1000, but nobody answered his call. The sound file selected
here will be played to John and then his call will be transferred to 'Last Destination'
extension)
(Select box)
• Loops:
How many times to dial all extensions again if nobody answers

(ex. John dials Ring group 1000, but nobody answers his call. If this option is set to '2',
all extensions will be dialed one more time before transferring his call to 'Last
Destination')
([0-9])
• Timeout:
How many seconds will all ring group extensions ring before the call is considered
unanswered
(ex. This option is set to 20. John dials ring group 1000. All Extensions will ring for 20
seconds before timeout occurs. Depending on whether 'Loop' option is set, all
extension will be rung again, or John will be transferred to 'Last Destination')
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([0-9])
• Dial Options:
Additional call options assigned to a ring group

(ex. To play music to ring group callers, set this field to 'm($CLASS)', where m = MOH
class e.g. m('default'). Please check details on the bottom)
([a-z])
• Ring Strategy:
This option regulates how extension in the Ring Group will be ringed.
Example:
• All - ring all extensions in the group

• Leastrecent - ring extension with least answered calls
• Round - ring each available extension
• Round Memory - like round, except we remember where we left off the last
ring pass
(Select box)
• Custom ringtone:
Set a custom ringtone for the phones which are in this ring group
(ex. More info can be found in this section: Call Filters & Blocking)
([0-9] [a-z])
• Record Calls:
Enable call recording service

(ex. Select 'Yes' to enable the service. All incoming/outgoing calls will be recorded. If
using call recording with many extensions, check server disk space from time to time.
Please see below for bit rates table).
(Option buttons)
• Silent:
Set whether call recordings should be announced to parties in a conversation.

(ex. If Silent=No, calling parties will hear a 'Recorded' or 'This call is recorded' message
before their conversation starts)
(Options buttons)
• Exit Digit:
64
Exit digits that transfers the call to the 'Exit Destination'
(ex. John dials ring group 1000. While all extensions are ringing, John presses the 'Exit
digit' set here (e.g. 9) and his call is transferred to the 'Exit Destination').
([0-9])
• Exit Extension:
PBXware MT extension to which the call is transferred once the user dials the 'Exit
Digit'
(ex. John dials ring group 1000. While all extensions are ringing, John presses the 'Exit
Digit' and his call is transferred to the 'Exit Destination' provided here (e.g. 2001))
([0-9])
• Last Destination:
Last destination to be dialed if none of the ring group extensions answer the call
(ex. John dials Ring group 1000, but nobody answers his call. Sound file selected under
'Announce' is played to John and his call is transferred to the extension number set
here).
([0-9])
• Last Destination is voicemail:
Choose whether you want calls to be redirected to the Last Destination or Last
Destination voicemail
(ex. Yes, No, N/A)
(Option buttons)
• Confirm Calls:
Chose whether the called number in the ring group list should be asked to accept or
refuse the call from ring group.
(ex. Yes, No, N/A)
(Option buttons)
• Confirmation Message:
Chose whether to play system default or some custom added sound asking if you want
to answer or reject the call
(ex. All sound files for this option should start with 'rg-announce')
(Select box)
65
• Call Answered Message:
Chose whether to play system default or custom sound file which is presented to user
when he accepts the call from ring group, but the call has already been answered by
someone else
(ex. All sound files for this option should start with 'rg-late-announce')
(Select box)
With continuous tone for 60 seconds
• wav49 = 84.5kb
• wav = 833.0kb
• gsm = 85.0kb
With continuous silent tone (without sound) for 60 seconds
• wav49 = 84.0kb
• wav = 827.0kb
• gsm = 84.0kb
Dial Options:

channel(s) until one answers. Use with care and don't insert this by default into all
• R - Indicate ringing to the calling party when the called party indicates ringing, pass no
audio until answered. This is available only if you are using kapejod's bristuff.
• m - Provide Music on Hold to the calling party until the called channel answers. This is
mutually exclusive with option 'r', obviously. Use m(class) to specify a class for the
music on hold.
• o - Restore the Asterisk v1.0 Caller ID behaviour (send the original caller's ID) in
Asterisk v1.2 (default: send this extension's number)
• M(x) - Executes the macro (x) upon connect of the call (i.e. when the called party
answers)
• h - Allow the callee to hang up by dialing *
command
• P(x) - Use the Privacy Manager, using x as the database (x is optional)
66
context.
• G(contextêxten^pri) - If the call is answered, transfer both parties to the specified
• A(x) - Play an announcement (x.gsm) to the called party.
• S(n) - Hang up the call n seconds AFTER the called party picks up.
• d: - This flag trumps the 'H' flag and intercepts any dtmf while waiting for the call to be
answered and returns that value on the spot. This allows you to dial a 1-digit exit
• D(digits) - After the called party answers, send digits as a DTMF stream, then connect
the call to the originating channel.
• L(x[:y][:z]) - Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms)
Only 'x' is required, 'y' and 'z' are optional. The following special variables are optional
for limit calls: (pasted from app_dial.c)
o + LIMIT_PLAYAUDIO_CALLER - yes|no (default yes) - Play sounds to the caller.
o + LIMIT_PLAYAUDIO_CALLEE - yes|no - Play sounds to the callee.
LIMIT_WARNING_FILE is not defined, then the default behavior is to announce
('You have [XX minutes] YY seconds').
• f - forces callerid to be set as the extension of the line making/redirecting the outgoing
call. For example, some PSTNs don't allow callerids from other extensions than the
ones that are assigned to you.
Departments
Departments
Departments section will list all the departments present on this <%PRODUCT%>
system, and give the ability to edit or add a new ones. Departments are used by Bicom
Systems gloCOM to sort extensions based on the department they belong to.
67
Add/Edit Department
Add/Edit Department
When you click on Add Department link or the edit button you will be presented with
this screen:
• Name:
Name of the department

(ex. Accounting)
([0-9][a-z])
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MT 3.8.5 Trunks
Contents
• 1 Trunks
o 1.1 Search
o 1.2 Add/Edit Trunk
o 1.3 Custom Trunks
Trunks
Trunks display screen
Trunks are a transmission line between two systems. This transmission is done using a
wide range of PSTN and VOIP technologies. This screen lists all system Trunks with
the following details:
• Name
This field tell us name of the trunk.

(ex. It can be TrunkName or IP address 192.168.1.6)
(Display)
• Provider
Provider template name.

(ex. SIP TRUNK)
• Channels
Maximum number of inbound/outbound channels available for the trunk.

e.g. 10/10
• Trunk Type
Type of a trunk
(ex. PSTN, VOIP etc.)
69
• Protocol
Protocol used by the trunk

(ex. SIP, IAX etc.)
• Click this button if you want to edit trunk configuration.
• Click this button in order to delete a trunk from the system
Search
The search bar filters trunks by name and provider.
Search filter
• Search
Search phrase
(ex. Provide a search phrase here and hit enter to filter the records)
([a-z][0-9])
• Name
Should the search filter be applied to trunk names.

(ex. Check the box to search trunk names)
(Check box)
• Provider
Should the search filter be applied to provider names.

(ex. Check the box to search provider names)
(Check box)
Add/Edit Trunk
When adding a new trunk, the first step requires 'Provider' and 'Device' selection.
70
TIP:
Although a new trunk can be created without it, it is preferred that the 'Provider'
template be created first under 'Settings: Providers'.
For basic Provider and Trunk HowTo check our HOWTO Create a Trunk
Add new trunk window
• Provider
Select a service provider template.

(ex. BT)
(Select box)
• Device
If the providers service requires a device in order to provide the service, this field will
become visible.
(ex. None, T100)
(Select box)
VoIP (SIP/IAX)
The second step of the trunk installation and trunk edit command, opens the following
options:
Edit trunks window
• Name or Number
71
Some providers require this field to be equal to the DID number (e.g.55510205); but if
connecting two systems, the IP address may be used as well.
(ex. 2554433, myvoiceboxlink)
([a-z] [0-9])
• Emergency trunk
Should emergency services (Police, Ambulance etc) be dialed through this trunk.
(ex. Dialing 911 will pass the call through this trunk)
(Option buttons)
• Peer Host
The IP of a peer host system sends the calls to.

(ex. 192.168.1.1)
(IP Address)
• Username
The username for authenticating with the service provider.

(ex. 2554433)
([0-9])
• Peer Username
The peer username for authenticating with the service provider.

(ex. 2554433)
([a-z] [0-9])
• Secret
The Secret/Password used for authenticating with the service provider.

(ex. 123456)
([a-z] [0-9])
• Peer secret
The peer secret/password used for authenticating with the service provider.
(ex. 123456)
([a-z] [0-9])
Advanced Options
72
General
These options are used frequently and required for normal trunk operation. Some of
these fields are pre-configured with the default values. It is not recommended to change
them.
General options
• User Type
The user's relationship to the system

Example:
• user - The trunk accepts incoming calls only

• peer - The trunk makes outgoing calls only
• friend - The trunk does both incoming and outgoing calls
(Select box)
• DTMF Mode (Dual Tone Multi-Frequency)
Trunk DTMF mode. A specific frequency (consisting of two separate tones) to each key
so that it can be easily identified by a microprocessor
Example:
• inband - inband audio (requires 64 kbit codec - alaw, ulaw)

• rfc2833 - default
73
• info - SIP INFO messages
(Select box)
• RFC2833 Compensate (1.2)
Compensate for pre-1.4 Asterisk DTMF transmission from another machine. You must
have this turned on or DTMF reception will work improperly.
(ex. Yes, No, N/A)
(Option buttons)
• Context
Contexts define a scope within PBXware. The trunk context cannot be modified and is
the same as the trunk name or number.
(ex. 2554433)
([a-z][0-9])
• Status
Set trunk status on the system.

(ex. Rather than deleting the trunk, you can disable it on the system level by selecting
'Not Activated'here)
(Select box)
• Qualify
Timing interval in milliseconds at which a 'ping' is sent to a host in order to find out its
status.
(ex. Set this field to 2000, for example. If more time than provided here is needed to
reach the host, the host is considered offline)
([0-9])
• VoiceMaster Trunk
Set whether this trunk leads to the VoiceMaster gateway.

(ex. If you have a VoiceMaster gateway and are creating this trunk to connect it with
this system, select 'Yes' here)
(Option buttons)
• Country
Country where the service provider resides.
74
(Select USA, for example, if the provider is from the United States)
(Select box)
• National Dialing Code
National dialing code used at the trunk destination.

(ex. For USA 1. For the United Kingdom and Germany 0)
(Select box)
• International Dialing Code
International dialing code used at the trunk destination.

(ex. For USA 011. For the United Kingdom and Germany 00)
([0-9])
• E164 Accepted
Does the trunk support dialing destinations in the E164 format.

(ex. Enabling this option will reformat any dialed number into the following form
COUNTRY_CODE + AREA_CODE + DIALED_NUMBER. For example, if you dial 55510205,
the system will dial 121255510205)
(Option buttons)
• Pass-thru mode
Pass the digits dialed without any conversion (E164, National, Area code).
NOTE: When active, 'Leave National Code and 'Local Area Code' will be disabled.
(ex. If this option is disabled, PBXware will convert all dialed numbers to E164 format
(COUNTRY_CODE + AREA_CODE + DIALED_NUMBER) and then make a call to
converted number. If this option is enabled, PBXware will directly call the
DIALED_NUMBER without making any number conversions)
(Option buttons)
• Leave National Code
In some countries, the national code is stripped automatically. If set to 'Yes', the
national code will not be stripped from the dialed number.
NOTE: Before setting this option to 'Yes', go to 'Settings: Servers' and enable this
option as well.
(ex. 035123456 will not be striped of 0)
(Option buttons)
75
• Local Area Code
Add the local area code to the dialed number, if required by the service provider. (By
default, the local area code is stripped when dialing).
(ex. The user dials 55510205, the local area code is 212. If the call goes through this
trunk, PBXware will dial 21210205)
([0-9])
• Prefix
Value added to all dialed numbers going over the trunk.

(ex. Prefix 5 + Dialed number 123 = System dials 5123)
([0-9])
• Outbound Caller ID
If the Caller ID is not set by the UAD, the value provided here will be used instead for
all outgoing calls.
(ex. 55599999)
([0-9])
• Allow ES Caller ID
Should ES (Enhanced Services) caller id be allowed over this trunk.

(ex. Any extension can set a custom caller id for each system trunk. With this option
enabled, that caller id will be used instead of the trunk outbound caller ID).
([0-9])
• Test number
This is the number that the trunk will try to call when Trunk Monitoring is turned on. If
the answer is not: ANSWER, BUSY, CANCEL, or NOANSWER, this call is considered non-
functional and the monitoring system will send a notification email.
(1009)
([0-9])
• Send PAI header
The P-Asserted-Identity contains the caller ID information for the call on the INVITE SIP
packet. This is the acceptable way to specify privacy information for calls. This field
enables you to send preferred PAI header using several variables:
• %CALLERIDNUM%
76
• %TENANT%
• %EXT%
• %TENANTEXT%
Authentication
Authentication settings
• Host
The IP address that the host trunk is connecting to.

Example:
Enter a host IP, 192.168.1.1, for example, or set 'dynamic' if the host is behind the
dynamic IP address.
([0-9][a-z])
• Authname:
Value assigned to the Digest username= SIP header

(ex. 2222)
([0-9][a-z])
• Auth
Authenticate for outbound calls to other realms.

(ex. user:secret@realm)
([0-9][a-z])
• Register
Method for registering to the remote server.

77
(ex. Providers may require a different form of registration to their server. You may
choose between 'registration not required', 'register with phone number', and 'register
with username').
(Select box)
• Register suffix
The service provider may request different registration methods for their services.
Select the proper method, as required by the provider.
(ex. 1234567)
([0-9])
• Insecure
Typically used to allow incoming calls while having a "friend" type entry defined with a
username and password
Example:
• Yes
• No
• very
• port
• invite
• port, invite
(Select box)
• From User
What to show when calling TO this peer FROM asterisk.

(ex. 152)
([0-9])
• From Domain
From domain data is required by some providers.

(ex. If your provider requires this information, provide the exact value here).
([a-z][0-9])
• user=phone in URI
If yes, ";user=phone" is added to uri that contains a valid phone number.

(ex. Yes, No, N/A)
(Option buttons)
78
• Outbound Proxy
Send outbound signaling to this proxy, not directly to the peer.

(ex. outbound.proxy.com)
([0-9][a-z].)
• Incoming IP addresses (new line seperated)
You may limit SIP traffic to and from this peer to a certain IP or network.
(ex. 10.1.1.9)
([0-9].)
IAX Specific Authentication Settings
• Encryption
Should encryption be used when authenticating with the peer.

(ex. Yes, No, N/A)
([a-z][0-9])
• Auth Method
Authentication method required by the provider.

(ex. md5)
([a-z] [0-9])
• RSA key
RSA authentication key

(ex. If the Auth Method is set to RSA, provide the RSA key here).
([a-z][0-9])
Network Related
These options set important network related values regarding NAT.
79
Network Related settings
• Transport:
Type of transfer protocol that will be used on PBXware.

UDP (User Datagram Protocol) - With UDP, computer applications can send messages,
in this case referred to as datagrams, to other hosts on an Internet Protocol (IP)
network without prior communications to set up special transmission channels or data
paths.
TCP (Transmission Control Protocol) - provides reliable, ordered, error-checked

delivery of a stream of octets between programs running on computers connected to
an intranet or the public Internet.
TLS (Transport Layer Security) - cryptographic protocol that provide communication

security over the Internet.[1] They use asymmetric cryptography for authentication of
key exchange, symmetric encryption for confidentiality, and message authentication
codes for message integrity.
Type: Checkbox
• Encryption:
This option enables or disables encryption in PBXware transport.

Options: Yes, No, N/A.
• Direct media
Should you allow RTP voice traffic to bypass PBXware.

NOTE: All enhanced services for the extension have to be disabled.
(ex. Some devices do not support this especially if one of them is behind a NAT).
(Options buttons)
• Direct RTP setup:
Here you can enable or disable the new experimental direct RTP setup. Setting this
value to yes sets up the call directly with media peer-2-peer without re-invites. Will
not work for video and cases where the callee sends RTP payloads and fmtp headers in
the 200 OK that does not match the callers INVITE. This will also fail if directmedia is
enabled when the device is actually behind NAT.
• Default IP
The IP address to be used until registration.

(ex. 192.168.1.1)
80
(IP Address)
• Use RTP source address for T.38 packets (1.2)
Use the source IP address of RTP as the destination IP address for T.38 packets if the
nat option is enabled. If a single RTP packet is received, Asterisk will know the external
IP address of the remote device. If port forwarding is done at the client side, then T.38
will flow to the remote device.
(ex. Yes, No, N/A)
(Option buttons)
Channels
Channels settings
• Incoming Limit
The number of simultaneous incoming calls that the trunk can handle.
(ex. 4 = four simultaneous incoming calls. Any additional calls will get a busy sound).
([0-9])
• Outgoing Limit
Number of simultaneous outgoing calls that the trunk can handle.

(ex. 4 = four simultaneous outgoing calls. Any additional calls attempting to use this
trunk will be rejected or will be redirected to other trunks depending on what is set in
the system/extensions).
([0-9])
• Busy Level
Number of concurent calls until user/peer is busy.
• E-mail on exceeded limit
Send an e-mail when the outgoing limit is reached, can be set to Yes, No or N/A.
(Option buttons)
81
• Outgoing Dial Options
Advanced dial options for all outgoing calls.

(ex. trT)
([a-z])
IAX Specific Channels settings
Channels settings
• Notransfer
Disable the native IAX transfer.

(ex. Yes, No, N/A)
(Option buttons)
• Send ANI
Should ANI ("super" Caller ID) be sent over this trunk.

(ex. Yes, No, N/A)
(Option buttons)
• Trunk
Use IAX2 trunking with this host.

(ex. Yes, No, N/A)
(Option buttons)
Codecs
Codecs are used to convert analog to digital voice signals and vice versa. These options
set preferred codecs used by the extension.
TIP:
If some of the desired codecs cannot be checked, go to 'Settings: Servers: Edit: Default
Codecs' and enable them under the 'Remote' group.
82
Codecs settings
• Disallow
The set that the codecs trunk is now allowed to use.

to 'Disallow All' and it cannot be modified).
(Read only)
• Allow
The set that the codecs extension is allowed to use.

Example:
Only the codecs set under 'Settings: Server' will be available to choose from

• ITU G.726 - 16/24/32/40 Kbps
• LPC10 - 2.5 Kbps
• H.263+ Video - Extension of H.263 that provides additional features that
improve compression over packet switched networks.
(Check box)
• Auto-Framing (RTP Packetization)
If auto framing is turned on, system will choose packetization level based on remote
ends preferences.
(ex. Yes)
(Option buttons
83
PSTN
• Name
Trunk name/number.
(ex. 032445231)
([a-z][0-9])
• Emergency trunk
Should emergency services (Police, Ambulance, etc) be called over this trunk.
(ex. Select 'Yes' in order to dial emergency services over this trunk).
(Option buttons)
• Channel(s)
Which card channels are used.

(ex. If channel 2 and 4 are used on your card, set '2, 4' here. If all four channels are
used set '1-4' here).
([0-9], [,-])
• Group
Every PSTN trunk has to belong to a group. Selecting any group will enable the trunk.
(ex. With most of the cards, this option is auto detected and set. If that is the case with
your card - do not change this field).
(Select box)
• FXS Kewlstarts
Signaling protocol for analog circuits that better detects far-end disconnects.
84
(ex. Select card channels to be monitored with it. For example '1, 4' or '1-4'. These
numbers should match the 'Channel(s)' field).
([0-9], [,-])
• Country
Destination of the trunk connection.

(ex. If your system is located in the USA, select USA here)
(Select box)
• E164 Accepted
Does the trunk support dialing destinations in the E164 format. Enabling this option
will reformat any dialed number into the following form
COUNTRY_CODE+AREA_CODE+DIALED_NUMBER.
(ex. If you dial 55510205, the system will dial 121255510205).
(Option buttons)
• International Dialing Code:
The international dialing code at the trunk destination.

(ex. For the USA 011, for the United Kingdom and Germany 00 etc.)
([0-9])
Advanced
Zapata
General
Zapata General settings
85
• Language
Default language.
(ex. us)
(Select box)
• Context
Contexts define a scope within the PBXware.

(ex. default)
([a-z][0-9])
• Status:
Trunk status
Example:
• Active
• Not Activated
(Select box)
• Signalling
Signaling method
Example:
• FXS Loopstart
• FXS Groundstart
• FXS Kewlstart
• FXO Loopstart
• FXO Groundstart
• FXO Kewlstart
• PRI CPE side
• PRI Network side
• BRI CPE side
• BRI Network side
• BRI CPE PTMP
• BRI Network PTMP
• Test number
This is the number which the trunk will try to call when Trunk Monitoring is turned on.
If the answer is not: ANSWER, BUSY, CANCEL, or NOANSWER, this call is considered
non-functional and the monitoring system will send a notification email.
(ex. 1009)
86
([0-9])
• Music On Hold
Select which class of music to use for music on hold. If not specified, the 'default' will
be used.
(ex. default)
(Select box)
• Mailbox
Define a voicemail context.

(ex. 1234, 1234@context)
([a-z][0-9)
• Group Method
([a-z][0-9])
This option specifies how to choose a channel to use in the specified group.
The four possible options are:
• g: select the lowest-numbered non-busy DAHDI channel (aka. ascending

sequential hunt group).
• G: select the highest-numbered non-busy DAHDI channel (aka. descending
sequential hunt group).
• r: use a round-robin search, starting at the next highest channel than last time
(aka. ascending rotary hunt group).
• R: use a round-robin search, starting at the next lowest channel than last time
(aka. descending rotary hunt group).
RX/TX
RX/TX settings
• RX Wink
Set timing parameters

Example:
87
• Pre-wink (50ms)
• Pre-flash (50ms)
• Wink (150ms)
• Receiver flashtime (250ms)
• Receiver wink (300ms)
• Debounce timing (600ms)
(Select box)
• RX Gain
Receive signal decibel.

(ex. If incoming sound is low and you cannot hear the other party well, set this option
to 2. That should increase incoming sound by 2 decibels).
([0-9])
• TX Gain
Transmit signal decibel.

(ex. If outgoing sound is low and the other party cannot hear you well, set this option
to 2. That should increase outgoing sound by 2 decibels).
([0-9])
PRI
PRI settings
• Switchtype
Set switch type

Example:
• National ISDN 2
• Nortel DMS100
• AT&T 4ESS
• Lucent 5ESS
• EuroISDN
88
• Old National ISDN 1
(Select box)
• PRI Dial Plan
Set the dial plan used by some switches

Example:
• Unknown
• Private ISDN
• Local ISDN
• National ISDN
• International ISDN
(Select box)
• PRI Local Dial Plan
Set the numbering dial plan for destinations called locally

Example:
• Unknown
• Private ISDN
• Local ISDN
• National ISDN
(Select box)
• PRI Trust CID
Trust provided caller id information.

(ex. Yes, No, N/A)
(Option buttons)
• PRI Indication
How to report 'busy' and 'congestion' on a PRI

Example:
• outofband - Signal Busy/Congestion out of band with RELEASE/DISCONNECT

• inband - Signal Busy/Congestion using in-band tones
(Select box)
• Network Specific Facility
89
If required by switch, select the network specific facility
Example:
• none
• sdn
• megacom
• accunet
(Select box)
Caller ID
Caller ID settings
• Outbound Caller ID
Caller ID set for all outbound calls where the Caller ID is not set or supported by a
device.
([0-9])
• Allow ES Caller ID
Should the ES (Enhanced Services) Caller ID be allowed over this trunk.

(ex. Any extension can set a custom Caller ID for each system trunk. With this option
enabled, that Caller ID will be used instead of the Trunk Outbound Caller ID).
([0-9])
• Caller ID (for analog or inbound)
CallerID can be set to 'asreceived' or a specific number if you want to override it.
NOTE: Caller ID can only be transmitted to the public phone network with supported
hardware, such as a PRI. It is not possible to set external caller ID on analog lines.
90
(ex. 'asreceived', 555648788)
([a-z][0-9])
• Use Caller ID
Whether or not to use caller ID.

(ex. Yes, No, N/A)
(Option buttons)
• Hide Caller ID
Whether or not to hide the outgoing caller ID.

(ex. Yes, No, N/A)
(Option buttons)
• Restrict CID
Whether or not to use the caller ID presentation for the outgoing call that the calling
switch is sending.
(ex. Yes, No, N/A)
(Option buttons)
• Use CallerID Presentation
switch is sending.
(ex. Yes, No, N/A)
(Option buttons)
• CID Signalling
Set the type of caller ID signaling

Example:
• bell - US
• v23 - UK
• dtmf - Denmark, Sweden and Netherlands
(Select box)
• CID Start
What signals the start of the Caller ID

Example:
91
• ring = a ring signals the start
• polarity = polarity reversal signals the start
(Select box)
• Call Waiting CID
Whether or not to enable call waiting on FXO lines.

(ex. Yes, No, N/A)
(Option buttons)
• Send CallerID After
Some countries, like the UK, have different ring tones (ring-ring), which means the
caller id needs to be set later on, and not just after the first ring, as per the default.
(ex. Yes)
(Select box)
Echo Canceler
Echo Canceler settings
• Echo Cancel
Level of enabled echo cancellation.

(ex. 128 (Yes))
(Select box)
• Echo Training
Mute the channel briefly, for 400ms, at the beginning of the conversation, cancelling
the echo. (Use this only if 'Echo Cancel' doesn't work as expected).
(ex. Yes, No, N/A)
(Option buttons)
• Echo Cancel When Bridged
92
Enable echo cancellation when bridged. Generally not necessary, and in fact
undesirable, to echo cancel when the circuit path is entirely TDM.
(ex. Yes, No, N/A)
(Option buttons)
Call Features
Call Features options
• Call Waiting
Whether or not to enable call waiting on FXO lines.

(ex. Yes, No, N/A)
(Option buttons)
• Three Way Calling
Support three-way calling. If enabled, the call can be put on hold and one is able to
make another call.
(ex. Yes, No, N/A)
(Option buttons)
• Transfer
Support call transfer and also enable call parking (overrides the 'canpark' parameter).
Requires 'Three Way Calling' = 'Yes'.
(ex. Yes, No, N/A)
(Option buttons)
• Can Call Forward
Support call forwarding.

(ex. Yes, No, N/A)
93
(Option buttons)
• Call Return
Whether or not to support Call Return '*69'. Dials last caller extension number.
(ex. Yes, No, N/A)
(Option buttons)
• Overlap Dial
Enable overlap dialing mode (sends overlap digits).

(ex. Yes, No, N/A)
(Option buttons)
• Pulse Dial
Use pulse dial instead of DTMF. Used by FXO (FXS signalling) devices.
(ex. Yes, No, N/A)
(Option buttons)
Call Indications
Call Indications settings
• Distinctive Ring Detection
Whether or not to do distinctive ring detection on FXO lines.

(ex. Yes, No, N/A)
(Option buttons)
• Busy Detect
Enable listening for the beep-beep busy pattern.

(ex. Yes, No, N/A)
(Option buttons)
94
• Busy Count
How many busy tones to wait before hanging up. Bigger settings lower the probability
of random hang ups. 'Busy Detect' has to be enabled.
• 4
• 6
• 8
(Select box)
• Call Progress
Easily detect false hangups.

(ex. Yes, No, N/A)
(Option buttons)
• Immediate
Should the channel be answered immediately or should the simple switch provide
dialtone, read digits, etc.
(ex. Yes, No, N/A)
(Option buttons)
Call Groups
Call Groups
• Call Group
Which group is allowed to pick up incoming calls by dialing *8. The default value is
empty.
(ex. 1, 1-4)
([0-9])
• Pickup Group:
Which groups are allowed to pick up calls by dialing *8. The default value is empty.
95
(ex. 1, 1-4)
([0-9])
FXS Channels
FXS Channels settings
• FXS Loopstart
Signals the far end that it wants the dial tone by shorting the leads.
(ex. default)
([0-9])
• FXS Groundstart
Signals the far end that it wants the dial tone by grounding one of the leads.
(ex. default)
([0-9])
Locality
Locality settings
• Country
Destination of the trunk connection
96
(ex. USA)
(Select box)
• E164 Accepted
Does the trunk support dialing destinations in E164 format.

COUNTRY_CODE+AREA_CODE+DIALED_NUMBER. For example, if you dial 55510205,
the system will dial 121255510205).
(Option buttons)
• Pass-thru Mode
If this option is enabled, the number which is dialed is passed through trunk without
modification.
(ex. Yes, No, N/A)
(Option buttons)
• National Dialing Code
The national dialing code at the trunk destination.

(ex. For the USA 1. For the United Kingdom and Germany 0 etc.)
([0-9])
• Leave National Code
NOTE: Before settings this option to 'Yes', go to 'Settings: Servers' and enable this
options as well.
(ex. John dials 121255510205. With this option enabled)
([0-9])
• International Dialing Code
International dialing code at the trunk destination.

(ex. For the USA 011. For the United Kingdom and Germany 00 etc.)
([0-9])
• Local Area Code
97
default, the local area code is stripped when dialing)
(ex. The user dials 55510205, the local area code is 212. If the call goes through this
trunk PBXware will dial 21210205).
([0-9])
• Write dialing code
Should the National and International prefix be written into the configuration files.
(ex. Enable this option if required by provider).
([0-9])
• Prefix
Value added to all dialed numbers going over the trunk.

(ex. Prefix 5, Dialed number 123, System dials 5123).
([0-9])
Other Zapata Options
• ADSI (Analog Display Services Interface)
Enable remote controlling of the screen phone with softkeys. (Only if you have ADSI
compatible CPE equipment).
(ex. Yes, No, N/A)
(Option buttons)
• Jitter Buffers
Configure jitter buffers. Each one is 20ms long.

(ex. 4)
([0-9])
98
• Relax DTMF
If you are having trouble with DTMF detection, you can relax the DTMF detection
parameters.
(ex. Yes, No, N/A)
(Option buttons)
• Fax Detect
Enable fax detection

Example:
• both
• incoming
• outgoing
• no
(Select box)
Span
Span settings
• Span number
Number of the span.

(ex. 1)
([0-9])
• Span timing
How to synchronize the timing devices

Example:
• 0 - do not use this span as sync source

• 1 - use as primary sync source
• 2 - set as secondary and so forth
99
([a-z])
• Line build out
Length of the last leg of the connection and is set to zero if the length is less than 133
feet
Example:
• 0 db (CSU) / 0-133 feet (DSX-1)

• 133-266 feet (DSX-1)
• 266-399 feet (DSX-1)
• 399-533 feet (DSX-1)
• 533-655 feet (DSX-1)
• -7.5db (CSU)
• -15db (CSU)
• -22.5db (CSU)
(Select box)
• Framing
How to communicate with the hardware at the other end of the line
Example:
• For T1: Framing is one of d4 or esf.

• For E1: Framing is one of cas or ccs.
(Select box)
• Coding
How to encode the communication with the other end of line hardware.
Example:
• For T1: coding is one of ami or b8zs

• For E1: coding is one of ami or hdb3 (E1 may also need crc)
(Select box)
• Yellow
Whether the yellow alarm is transmitted when no channels are open.

(ex. Yes, No, N/A)
(Option buttons)
Dynamic Span
100
Dynamic Span settings
• Dynamic span driver
The name of the driver (e.g. eth).

([0-9][a-z])
• Dynamic span address
Driver specific address (like a MAC for eth).

([0-9][a-z])
• Dynamic span channels
Number of channels.
(6)
• Dynamic span timing
Sets timing priority, like for a normal span. Use '0' in order to not use this as a timing
source, or prioritize them as primary, secondary, etc.
(0)
FXO Channels
FXO Channels settings
• FXO Loopstart
Channel(s) are signaled using FXO Loopstart protocol.
• FXO Groundstart
101
Channel(s) are signaled using FXO Groundstart protocol.
• FXO Kewlstart
Channel(s) are signaled using FXO Kewlstart protocol.
PRI Channels
PRI Channels settings
• D-Channel(s)
For example, every ISDN BRI card has 1 D- (control) channel.

(ex. 1)
([0-9])
• B-Channels(s)
For example, every ISDN BRI card has 2 B- (data) channels.

(ex. 2)
([0-9])
Custom Trunks
Custom Trunks
Custom trunks are used in rare cases when PBXwares trunk setup features are not
providing enough flexibility for some provider configurations. In such cases one will
create a custom trunk and then change the dialplan in the trunks-in.conf file.
For information on how to modify asterisk diaplan, please check sites like www.voip-
info.org
102
Add Custom Trunk
When you click on Add Custom Trunk button, you will get following screen:
Edit Custom Trunks options
It is the same screen when adding or editing the current custom trunk.
• Trunk
Name of the trunk which will be visible in PBXware GUI.

(ex. Global Trunk)
([0-9][a-z])
• Context
Context by which the system sees this trunk and which will be used when editing the
dialplan.
(ex. Set gl_trunk for example, and when sending calls to this trunk, you would use this
to refer to the given trunk).
([0-9][a-z])
103
MT 3.8.5 DID
Contents
• 1 DIDs Main Screen
o 1.1 Search
o 1.2 CSV Upload/Download
• 2 Add DID
• 3 DIDs in Slaves
o 3.1 Edit DID
o 3.3 Operation Times
DIDs Main Screen

DIDs are used to point all incoming calls (that come over trunks) to specific system
destinations.
DIDs are configured on master tenant and assigned to a slave tenant on which they are
going to be used. This screen lists all system DIDs on all tenants with the following
details
104
DIDs
• 'Tenant:
Which tenant is using a DID.

(ex. Slave #3)
(Display)
• DID/Channel:
DID number or a PSTN channel slot number.

(ex. 1000/1)
(Display)
• Provider:
Provider name
(ex. SIP TRUNK)
(Display)
• Trunk:
Trunk used by a DID

(ex. 192.168.1.6/TrunkName)
(Display)
• Destination:
Trunk destination and destination network number

(ex. Network User - 5000)
(Display)
• Status:
DID status
(Display)
• Edit
Edits the DID configuration

(ex. Click to edit the DID configuration)
(Button)
• Delete
Deletes a DID from the system

(ex. Click to delete a DID from the system)
105
(Button)
Search
By selecting the 'Search' Command, the search menu will be displayed. Searches can be
done by DID value, Trunk name, Provider name, and Destination value
Search
• Search:
Search phrase
([a-z][0-9])
• DID:
Should a search filter be applied to DID values

(ex. Check the box to search DID values)
(Check box)
• Destination:
Should a search filter be applied to DID destinations

(ex. Check the box to search DID destinations)
(Check box)
• Country:
Should a search filter be applied to the country field

(ex. Check the box to search countries)
(Check box)
• State:
Should a search filter be applied to the state field

(ex. Check the box to search states)
(Check box)
• City:
Should a search filter be applied to the city field

(ex. Check the box to search cities)
(Check box)
106
• Area Code:
Should a search filter be applied to the area code field

(ex. Check the box to search area codes)
(Check box)
CSV Upload/Download
CSV Upload/Download
Here you can create multiple DIDs on the fly by uploading a .csv file with DID details,
or you can download csv for present DIDs and review or edit them.
To upload, click on the 'Browse' button, select a .csv file from your computer, click
'Open' and then the 'CSV Upload' button.
CSV file must be in the following format:
tenant,nr1,nr2,dest,ext,replace_cid,trunk,status,strip_N_digits,queue_
priority,codec,E164nr1,E164nr2,area_code
These fields are:
• tenant - Which tenant will use this DID

• nr1 - DID/Channel (start)
• nr2 - DID/Channel (end)
• dest - The destination field can contain one of the following values (case
insensitive and PBXware MT will tolerate white spaces if there are more than
needed i.e. 'Multi User'):
o Extension
o Multi User
o IVR
o Queues
o Voicemail
o Remote Access
o Conferences
o FAX to E-mail
o Phone Callback
• ext - E-mail/Extension/Value should contain destination value, except if one of
the following destinations is selected:
o Remote Access - must be one of the following values (case insensitive):
• Destinations
• Voicemail
• Agent Login
• replace_cid - Replace CallerID

107
• trunk - The trunk must contain a valid name of the trunk associated with DID
(case sensitive)
• status - Status
o 0 - Active
o 1 - Not Assigned
• strip_N_digits - Strip N digits
• queue_priority - Queue Priority
• codec - Force Codec. One of following values are acceptable by PBXware MT
(case sensitive):
o ulaw - G.711 ulaw
o alaw - G.711 alaw
o g723.1 - G.723.1
o g726 - G.726
o g729 - G.729
o gsm - GSM
o ilbc - iLBC
o speex - Speex
o lpc10 - LPC10
o h261 - H.261 Video
o h263 - H.263 Video
o h263p - H.263+ Video
• E164nr1 - E.164 number (start)
• E164nr2 - E.164 number (end)
• area_code - Area Code
Required fields:
• DID/Channel (start)
• Trunk
• Destination
• E-mail/Extension/Value
All other fields also must be defined in csv but they can contain empty values.
Add DID
To add a DID on the system you need to be on the master tenant where you can add and
set DID options in regard to the tenant which will use it, the trunk from which the DID
is dialed, etc. Clicking on 'Add DID' will open standard DID options
Add DID
• Tenant:
108
Select which tenant will use this DID.
(ex. Trunk #1)
(Select box)
• Trunk:
Select which trunk the DID will pick up calls from.

(ex. If you select '2554433' for example, DID will wait for any incoming calls
over that trunk and then will pass the call further based on the settings below).
(Select box)
• DID/Channel (start):
Provide a DID number here (e.g. 55510205)

(ex.If the selected 'Trunk' is PSTN or VOIP, set the line number here).
([0-9])
Advanced Options
A click on the 'Advanced Options' button will open more detailed DID options
Advanced Options
• Status:
Set the DID status on the network.

(ex. Rather than deleting the DID you can deactivate it by selecting 'Not
Activated' and restore it back with 'Active')
(Select box)
• Range:
Some providers offer a range of numbers over a single trunk. Set whether this
DID should be used to transfer a range of numbers to some PBXware MT
destination.
109
(ex. John has bought 10 DID numbers from a provider (55510205 - 55510215)
and wants all calls coming from these to be transferred to lobby queue. He needs
to set this options to 'Yes'. Set 55510205 to (start), 55510215 to (end) fields. Set
'Destination'='Queues' and 'Value'=Queue number(e.g. 1000)).
(Option buttons)
• DID/Channel (end):
Some providers offer a range of numbers over a single trunk. Set whether this
DID should be used to transfer a range of numbers to some PBXware MT
destination.
(ex. John has bought 10 DID numbers from a provider (55510205 - 55510215)
and wants all calls coming from these to be transferred to the lobby queue. He
needs to set 'Range'='Yes'. Set 55510205 to (start) and 55510215 to this field.
Set 'Destination'='Queues' and 'Value'=Queue number(e.g. 1000)).
(Option buttons)
• Force Codec:
Force a codec to all calls going over this DID

(ex. Select G.711 ulaw from the list to force all calls going over this DID to this
codec)
([a-z][0-9])
• E.164 number (start):
A DID number in E.164 format 'INTERNATIONAL PREFIX + AREA CODE +

PHONE NUMBER' ( 1 212 555 9876 ). If provided here, this number will be
used by 'PBXware MT: Networks' and will be dialed over the Internet rather
than the PSTN trunk. If the 'Range' field is set to 'Yes', provide the DID/Channel
(start) number in E.164 format here
(ex. If your DID number = 5559876, and you live in NewYork/USA, your E.164
number is 12125559876)
([0-9])
• E.164 number (end):
A DID number in the E.164 format 'INTERNATIONAL PREFIX + AREA

CODE + PHONE NUMBER' (1 212 555 9876). If 'Range' field is set to 'Yes',
provide the DID/Channel (
(ex. If your DID number = 5559876, and you live in NewYork/USA, your E.164
number is 12125559876)
([0-9])
• Country:
Select the country that this DID number belongs to

(ex. If the DID number is in the USA format (e.g. 1212****), select USA here)
([a-z][0-9])
110
• County:
The state that the DID number belongs to

(ex. If the DID number is in the USA format (e.g. 1212****), 212 is in New
York, so type NY here)
([a-z][0-9])
• City:
The city that the DID number belongs to

(ex. If the DID number is in the USA format (e.g. 1212****), 212 is in New
York, so type New York here)
([a-z][0-9])
• Area Code:
The area code that the DID number belongs to

(ex. If the DID number is in the USA format (e.g. 1212****), type 212 (the area
code for New York))
([0-9])
DIDs in Slaves
When you create a DID on master tenant, you can set its function in the slave that it is
assigned to.
DIDs in Slaves
• DID/Channel:
DID number or a PSTN channel slot number.

(ex. 1000/1)
(Display)
• Provider:
Provider name
(ex. SIP TRUNK)
(Display)
• Trunk:
111
Trunk used by a DID
(ex. 192.168.1.6/TrunkName)
(Display)
• Destination:
Trunk destination and destination network number

(ex. Network User - 5000)
(Display)
• Status:
DID status
(Display)
• Edit
Edits the DID configuration

(ex. Click to edit the DID configuration)
(Button)
Edit DID
When you click on the Edit button beside a DID, you will be able to edit that DID
settings, its destination, etc.
Edit DID
• Destination:
PBXware MT destination to which DID will transfer all calls. (Extensions,

IVRs, Queues, Voicemails, Remote Access, Conferences, and even to Fax to
Email service)
(ex. Extensions, IVRs, Queues, Voicemails, Remote Access, Conferences, or
Fax to Email)
(Select box)
• Value:
Destination extension to which all DID calls will be transferred
112
(ex. If 'Destination'='Extension', set the extension number here(e.g. 1002). In
case 'Destination'='IVR', set the IVR extension number here)
([0-9])
TIP: If using 'Fax to email', set the 'Value' field to an email address
([email protected]) or extension number (1002). If an extension number is used, the
fax will go to the email address associated with the extension. The email which arrives
to the specified address will contain both TIFF and PDF versions of the FAX
Advanced Options
Advanced Options
• Operation Times:
Set the DID operation time

For more explanation, click Operation Times
(Option buttons)
• Service Plan:
This option is used so you can bill inbound calls to IVRs, Conferences, Queues,
and Voicemail. If these are selected as Destination, they will be billed as 'Local
destinations' from Service Plan.
If Extension/Multi-Extension is selected, call will be billed as per incoming
price set for E.164 number in DID. Balance on Extension could be set as + or -
value so you can decrease or increase balance for that extension when it receives
the call. For other Destination Types Service Plan is disabled
(ex. National Service Plan)
(Select box)
• Greeting:
Greeting which is played when calling this DID

When uploading a greeting for DID, the name of the sound file must start with
'greeting-did-'
(Select box)
• Strip N digits
113
Here you can set the number of digits which will be stripped from the beggining
of the incoming call.
(ex. 3)
([0-9])
• Replace Caller ID:
Replaces the caller id with the custom data provided here. This is used when you
want all incoming DID calls to have this value displayed as Caller ID
information. Along with the custom data, you can use the '%PRODUCT%'
variable, which displays the calling party phone number.
NOTE: Please make sure you enter this information as it is written down,
otherwise it will not work properly.
(ex. Providing a 'USDID' here will display 'USDID' on your phone display for
all calls coming through this DID. Providing 'USDID %PRODUCT%', will
display 'USDID 55510205' on your phone display, where 55510205 is the
calling party's phone number).
([a-z][0-9] [%CALLERID%])
• Queue Priority:
Set the Queue priority

(ex. If this DID redirects all calls to the queue, set '1' here to give all calls over
this trunk the highest queue priority).
([a-z][0-9])
• Custom Ringtone
If you are directing calls to an extension on which a supported UAD is

registered, you could set a Custom Ringtone with which the phone will ring.
Read more on this in the Custom Ringtone How-To
(ex. <Simple-3>)
([a-z][0-9])
• Record Call:
If this option is turned on the incoming call on this DID will be recorded
(ex. When this option is turned on, a complete call will be recorded, whenever it
goes or whatever happens to it)
(Option buttons)
Operation Times
Set the system open/closed times. Depending on the time when the call is received, the
call can be redirected to different PBXware MT destinations.
114
Operation Times
Enable operation times

(ex. Yes, No)
(Option buttons)
PBXware MT extension to which all calls are redirected during the closed time
hours
(ex. 1000)
([0-9])
• Greeting:
Greeting sound file played to callers during the closed times

(ex. greeting-***)
(Select box)
Range Destinations: Redirects all calls received during non-working hours (e.g.
weekend) to the PBXware MT extension provided here.
Open dates: Sets the working hours during which the DID is to redirect calls as set in
the DID Add/Edit window. If any call is received during the hours not set here, 'Range
Destination' is checked, and if they do not apply, the call is redirected to 'Default
Destination'.
Closed dates: Sets the specific date when all calls are redirected to 'Default
Destination'. If 'Destination' field in the Closed dates is set, call will not go to 'Default
Destination' but to this number.
115
MT 3.8.5 Conferences
Contents
• 1 Conferences
o 1.1 Search
o 1.2 Add Conference
o 1.3 Groups
Conferences
Conferences allow two or more participants to communicate with each other at the same
time using voice, video, or both. Default limit for active conference members in
PBXware is 256 users in all conferences, meaning that you can have a single or multiple
conferences but their total active users count cannot exceed 256.
This screen lists all system conferences with the following details
• Name:
Conference name
(ex. Sales, Development)
(Display)
• Conference Number:
Conference system or network number

(ex. 2255)
(Display)
• Edit
Edits the Conferences configuration

(ex. Click to edit Conference configuration)
(Button)
116
Conferences
• Delete
Deletes a Conference from the system

(ex. Click to delete a Conference from the system)
(Button)
Search
done by Name and Number
Search
• Search:
Search phrase
([a-z][0-9])
• Name:
Should the search filter be applied to conference names

(ex. Check the box to search conferences names)
(Check box)
• Number:
Should the search filter be applied to conference numbers

(ex. Check the box to search conference numbers)
(Check box)
Add Conference
Clicking on the 'Add Conference' button will allow you to add new conference.
Add Conference
• Conference Name:
117
Unique Conference identifier/name
This name will be displayed once the 'Conferences' menu is selected
(ex. Main)
([a-z][0-9])
• Conference Number:
Unique Conference PBXware MT number

This number is to be dialed in order to access the conference
(ex. 106)
([0-9])
• Conference Group:
Here you can select a conference group that you have made before. In this group
you can define users permissions and conference options.
(ex. Staff)
(Select box)
Advanced Options
Clicking on the 'Advanced Options' button will open more detailed conference options.
Advanced Options
• Maximum number of users:
This is the maximum number of users permitted in Conference

(ex. 10)
([0-9])
• Conference PIN:
Conference PIN number. If set, this PIN will have to be provided by all callers
before entering the Conference.
(ex. John dials the sales conference (1001), but this conference has this field set
to 9576. John is asked to provide the conference PIN. If valid PIN is provided,
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John will enter the conference, or his call will be rejected after 3 invalid PIN
entries).
([0-9])
• Conference Admin PIN:
When users enter an Admin PIN, they dynamically gain admin rights for the
conference that they are dialing in.
(ex. John is the operator but he needs to enter the conference as an admin. Since
he does not have admin privileges set, when entering conference he will use
admin PIN to dynamically gain those privileges).
([0-9])
• Conference Marked User PIN:
When users enter Marked PIN, they dynamically gain marked user rights for the
conference that they are dialing into.
(ex. John is the operator but he needs to enter the conference as a Marked user.
Since he does not have Marked user privileges, when entering conference he
will use Marked user PIN code to dynamically gain those privileges).
([0-9])
• Rings to answer:
Number of rings played to the caller before a call is allowed to enter the
Conference
(ex. Rather than just 'falling' into Conference, it is recommended to set the
number of ring sounds played to the caller)
([0-9])
• RTP Delay [sec]:
Delay time in seconds inserted before the conference operator answers. This
delay solves the 'half-played' greeting file problem. Keep this value set between
1-3 for optimal performance.
NOTE: The 'half-played' greeting file problem usually exists on VOIP trunks.
(ex. John enters the sales conference (1001) and hears '...e currently the only
user in this conference' message. This message is only partial and can confuse
conference members. Set this field to 1 (or higher, depending on how much is
not heard) to play the message in full length).
([0-9])
• Conference color
Determines which color will be used for this conference in Bicom Systems
GLOCOM
(ex. Black)
(Select box)
119
Groups
Conference Groups are used to make groups of users that will have selected permissions
for conferences using them.
Groups
• Group Name:
Name of the conference group

(ex. Staff)
(Display)
• Users:
User Extensions that are added to this group

(ex. 101, 108, 109)
(Display)
• Edit
Edits the Conference Group configuration

(ex. Click to edit Conference Group configuration)
(Button)
• Delete
Deletes a Conference Group from the system

(ex. Click to delete a Conference Group from the system)
(Button)
Add Conference Group
When you click on Add Conference Group button, you will be able to add a new
Conference Group.
120
Add Conference Group
• Announce user join/leave w/review: (ast 1.4)
If this option is checked/enabled, all new conference members will be asked to

say their name and press the '#' key before they enter the conference. After the
user presses the '#' key he will be presented with the following menu:
• 1 - press 1 to accept your name and enter the conference

• 2 - press 2 to listen to your name
• 3 - press 3 to re-record your name
This name will be recorded and played to other conference members when caller
joins/leaves the conference
(ex. John dials 1001 sales conference and is asked to say his name and to press
the '#' key. Of course, he says 'John' and presses the '#' key. After that he is
presented with a menu where he chooses to accept his name and enter the
conference. At the same time, all conference members will hear 'John has
entered the conference'. When John leaves, all conference members will hear
'John has left the conference' message)
(Check box)
• Announce user join/leave:
If this option is checked/enabled, all new conference members will

(ex. John dials 1001 sales conference and is asked to say his name and to press
the '#' key. Of course, he says 'John' and presses the '#' key. At the same time, all
conference members will hear 'John has entered the conference'. When John
leaves, all conference members will hear a 'John has left the conference'
message).
(Check box)
• Quite mode:
If this option is checked/enabled, conference members will not hear the enter
and leave sound
(ex. Check this option if you don't want to hear the join/leave sound)
(Check box)
• Record conference:
This option will record the conference so that you can download the record file.
121
NOTE: You can download the recorded file in Conferences: CDR's, by
selecting a conference and clicking on the Listen button
(ex. Yes)
(Check box)
• Enable music on hold:
Enable MOH (Music On Hold) if there is a single member in a conference

(ex. John enters the conference and he is the only one there. Don't let John feel
alone. Enable this feature to play MOH class music files until someone else
joins the conference :))
(Check box)
• Present menu:
Returns the Conference options once * is dialed while in the conference

(ex. John enters the sales conference and dials '*'. Conference options are played
back (e.g. 'Please press 1 to mute/un-mute yourself'))
(Check box)
• Announce number of participants:
Announces the number of conference participants to a new conference member.

There is currently only 1 other participant in the conference
(ex. Sales conference has this option enabled and is currently empty. John enters
the conference and hears 'You are currently the only person in this conference'
payed back to him).
(Check box)
• Set talker detection
Enable the talker detection which is sent to manager interface and conferences
List.
(Check box)
• Close the conference when last marked user exits
Closes the conference once the last marked user exits, no matter how many
participants are still active in the conference conversation; their calls get
immediately dropped.
(ex. John (marked user) enters the sales conference. This conference has this
option enabled and there are 3 more members participating in the conference
conversation. As soon as John leaves the conference, all other conference
members will have their calls dropped and will no longer be able to talk to each
other).
(Check box)
• Wait the marked user before allowing anyone to talk
122
Disables the conference conversation until the marked user enters the
conference.
(ex. This option is enabled for sales conference. There are 3 members waiting
for John (marked user) to enter the conference. These 3 members will hear
nothing, and will not be able to talk to each other until the John enters the sales
conference).
(Check box)
• Dynamic Conference:
Dynamically adds a conference

(Check box)
• Dynamic Conference with PIN prompt:
Dynamically add a conference with the ability for the first member to set the
conference PIN number
(Check box)
• Set talker optimization:
Treats talkers who aren't speaking as being muted, meaning no encode is done
on transmission and received audio that is not registered as talking is omitted
causing no buildup in background noise
(ex. Yes)
(Check box)
• Enable Noise Reduction
This option will enable noise reduction for users inside the conference room.
(Check box)
Users
Users settings
There are two options for enabling users to join specific conference room. You can
either enable all users by selecting check box for Enable All Users option or you can
manually add users by clicking + sign in upper right corner of the Users table. This will
open Conference Groups: Add User pop-up window where you can perform search by
Extension number, Name or e-mail address. After your search results are displayed,
simply click on user that you want to add to Users table.
123
Add User window
• Enable All Users:
With this option you will allow all users to enter a conference
(ex. Yes)
(Check box)
• Extension:
The extension number of the user that you have added to the group
(ex. 233)
(Display)
• Name:
The name of the user that you have added to the group
(ex. John Crichton)
(Display)
• Admin
Sets the admin conference mode. If this option is enabled, conference calls
coming from the extension will be treated with admin privileges
(ex. No)
(Check box)
• Marked
Sets the marked conference mode. If this option is enabled, conference calls
coming from the extension will be treated with less than admin, but higher than
regular conference participant privileges
(ex. No)
(Check box)
• Talk-Only
Sets the talk only conference mode. If this option is enabled, conference calls
coming from this extension will be allowed to talk only (no voice will be heard
on the UAD/Phone)
(ex. No)
(Check box)
• Lstn-Only
Sets the listen only conference mode. If this option is enabled, conference calls
coming from the extension will be allowed to listen only (no voice will be sent
from the UAD/Phone)
(ex. No)
124
(Check box)
• Exit-#
If this option is enabled, the user will be allowed to exit the conference by
dialing the '#' key
(ex. No)
(Check box)
• Exit-Digit
If this option is enabled, the user will be allowed to exit the conference by
dialing any digit
(ex. No)
(Check box)
MT 3.8.5 IVR
IVRs (Interactive Voice Response) are automated answering machines which guide
callers to their destination by providing a number of choices and waiting for caller to
make a selection through DTMF tones via device keypad.
Contents
• 1 IVR
o 1.1 Search
o 1.2 Add/Edit IVR
o 1.3 Destination Types
o 1.5 General
o 1.6 General Settings
o 1.7 Greeting Options
o 1.9 Ringing Type
o 1.10 Local Dialing
o 1.11 Permissions
• 2 Multi-digit IVR codes

o 2.1 Search
125
o2.2 Add/Edit Access Code
• 3 PIN-based IVR
o 3.1 Search
o 3.2 Upload
o 3.3 Add/Edit PIN
• 4 IVR Tree
o 4.1 Add/Edit IVR
o 4.2 Actions & Selections
• 5 IVR Tree Graph
IVR
IVR settings screen
This screen lists all system standard DIDs with the following details:
• Name
IVR name
(ex. Welcome)
(Display)
• Number
IVR network number.

(ex. 1010)
• Edits the IVR configuration
(ex. Click to edit IVR configuration)

(Button)
• Deletes an IVR from the system
126
(ex. Click to delete an IVR from the system)
(Button)
Search
IVR search
• Search
Search phrase
(ex. Provide a search phrase here and hit enter to filter the records).
([a-z][0-9])
• Name
Should search filter be applied to IVR names

(ex. Check the box to search IVR names).
(Check box)
• Number
Should search filter be applied to IVR numbers.

(ex. Check the box to search IVR numbers).
(Check box)
Add/Edit IVR
Click on the 'Add/Edit' button will open standard IVR options
TIP:
Make sure to create a greeting sound before adding a new IVR. You may create one by
dialing '*301' from your UAD/Phone or by uploading a custom sound file from your
computer through 'System: Sounds'.
127
Edit IVR
• Name
Unique IVR identifier/name.

(ex. This name will be displayed once IVRs are accessed).
([a-z][0-9])
• Number
Unique network IVR number.

(ex. This number is to be dialed in order to access the IVR).
([0-9])
• Greeting
Greeting sound file.

Example:
Once a user enters the IVR, a greeting with instructions is played(e.g. 'Welcome.
For Sales Press 1...'). Select the greeting file played by this IVR here.
NOTE: Greeting file name must start with 'greeting-***'. To record a custom
greeting message dial '*301' from your extension. A newly recorded greeting file
will have the current date stamp in the title(e.g. 'greeting-Apr-14-2006-16-32').
(Select box)
• IVR Type
Set the proper IVR type.

(ex. PBXware works with two type of IVRs: Single and Multi digit ones. Single
digit IVR is used for small range of options(0-9). Multi digit IVR support
between(10-9999999999) and is shared with all Multi Digit IVRs).
(Select box)
• Destination
Set the proper destination for each digit pressed.
128
(ex. Once a greeting message(e.g. 'Press 1 for Sales') is played to user, provide
the valid destination where the call is to go to once 1 is pressed. If John from
sales department is to be dialed, select 'Extension' in this field. If you wish to
provide additional options to caller, you can point him to another IVR with its
set of options by selecting 'IVR' here).
(Select box)
• Extension
This field further describes the 'Destination' field. In case 'Remote Access' or
'Queue' are selected under 'Destination', a predefined option will be available for
selection under this option.
(In the example above we have set the PBXware destination. In this option we
set which destination part is to be dialed exactly. If 'Destination'='Extension',
provide the extension number here. If 'Destination'='IVR' provide the IVR
number here etc.).
([0-9])
Destination Types
• IVR
Destination for this selection is IVR with number entered into Extension field.
(ex. 401)
([0-9])
• Queue
Destination for this selection is the Queue which is selected from the Extension
select box.
(ex. Queue 1)
(Select box)
• Conference
Destination for this selection is the Conference with number entered into the
Extension field.
(ex. 500)
([0-9])
• Extension
When the Extension is selected, the destination for this selection is the Extension
with number entered into the Extension field.
(ex. 198)
([0-9])
• Voicemail
129
With this option selected, you can leave voicemail for a specified extension.
(ex. 198)
([0-9])
• Directory
With this option selected, you will have the ability to dial an extension by
entering the first three letters of the extension's last or first name, if it is
provided. Dialing by the first or last name depends on the selection in the
'Options' menu for that selection.
• Remote Access
This option enables you to remotely access one of four available types of
destinations:
Agent Login - enables remote login as an agent
Destinations - enables one to dial any destination
Voicemail - remotely login to Voicemail
(Select box)
• Fax to E-mail
When the user chooses this option, his fax can then be sent as an E-mail to the
number provided in the Extension field.
(ex. 222)
([0-9])
• Call External Number
When Call External Number is selected, destination for this selection is an

external number that you will enter in the Extension field.
(ex. 004412345678)
([0-9])
When you click on the Options button you will get the following window with two or
three options, depending on your destination selection.
IVR Options
• Caller ID
Overrides the incoming Caller ID with custom information.

(ex. Sometimes, it is useful to know from which IVR the call is coming from.
By settings 'Lobby IVR' here, all calls coming through this IVR will display
130
'Lobby IVR' on phone display. To show the actual phone number along with our
data use '%CALLERID%' with our text(e.g. 'Lobby IVR %CALLERID%'). This
will display 'Lobby IVR 55528790' on our phone display, where 55528790 is the
phone number of the person calling us).
([a-z][0-9], %CALLERID%, %CALLERIDNUM%, %CALLERIDNAME%)
• Language
Language used for this choice

(ex. us)
([a-z])
• Queue Priority
If the selected destination is a queue, this is where you set your priority in
regards to other callers in that queue, with 1 being the highest priority.
(ex. 5)
([0-9])
• Read extension number
If this option is selected, when the IVR selection is Directory, the system will
spell the extension number after the extension name.
(Check box)
• Search for name
Here you can select whether to search the directory by first or last name.
(ex. Last Name)
(Select box)
Advanced Options
General
IVR Edit window
131
• Status
Rather than deleting the IVR, set its status to 'Off'. This will make the IVR
inactive and all calls will be transferred to 'Operator Extension'.
(ex. Lobby IVR has this option set to 'Off'. John dials this IVR number (e.g.
1003) but instead of IVR instructions, his call will be transferred to 'Operator
Extension').
(Option button)
• Operator extension
Provide the operator extension to which all calls will be redirected to if 'IVR
Status' = 'Off'.
(ex. Lobby IVR has the 'Status' set to 'Off'. John dials this IVR but instead of
IVR instructions, his call will be transferred to the extension number provided
here).
([0-9])
• Disable CallerID (PIN-based IVR only)
This option disables CallerID but only for PIN-based IVRs.

(ex. Yes)
(Option button)
General Settings
IVR General Settings
• Response Timeout
Time period in seconds during which an IVR option must be dialed by the user.
(ex. John enters the Sales IVR and hears the instructions. If this field is set to '4',
John will have 4 seconds to dial an IVR option).
([0-9])
• RTP Delay [sec]
Delay time in seconds inserted before the IVR greeting message is played. This
solves the 'half-played' file problem. Keep this value between 1-3.
(ex. User A enters the IVR and hears a message '..me. For sales press 1' and
doesn't understand. Set this field to 1 so that a 1 second pause is added before
the message is played. Now, when user A enters the IVR he will hear 'Welcome.
For Sales press 1').
([0-9])
• Digit Timeout
132
Timeout in seconds during which a new digit must be dialed. This option is used
with Multi-digits IVR
(ex. John has entered the IVR and wants to dial option 25. If 1 is provided in this
field, John will have 1 second to dial number 2, and additional 1 second to dial
number 5. If the time exceeds, and John hits 5 too late, IVR will assume that
John has dialed option 2 instead of 25).
([0-9])
• Rings to Answer
Number of rings played to caller before a call is allowed to enter the IVR.
(ex. Rather than just 'falling' into IVR, it is recommended to set the number of
ring sounds played to caller).
([0-9])
IVR Greeting settings
Greeting Options
• Play Greeting
Number of times the greeting message is played to the caller. If there is no

response from the calling party within this time, the call is disconnected.
(ex. John enters the sales IVR and hears the IVR options. If John does not dial
one of the options, the IVR options sound file will be played again, a number of
times set in this field, before the call gets transferred to 'Timeout Extension').
([0-9])
• Timeout Extension
The extension number to which the IVR call will be transferred if there is no
response from the user during the 'Play Greeting' time period.
(ex. John enters the sales IVR and hears the IVR options. If John does not dial
one of the options, the IVR options sound file will be played again, a number of
times set in 'Play Greeting' field, before the call gets transferred to extension
number provided here).
([0-9])
Operation Times
Set the IVRs open/closed times. Depending on the time when the call is received, the
call can be redirected to different PBXware destinations.
133
IVR Operation Times
• Operation Times

(ex. Yes, No)
(Option buttons)
PBXware extension all calls are redirected to during the closed time hours
(ex. 1000)
([0-9])
• Greeting

(ex. greeting-***)
(Select box)
Description of destinations follows in this priority order:
• Open dates: Sets the working hours during which DID is to redirect calls as set
in DID Add/Edit window. If any call is received during the hours not set here,
'Custom Destination' are checked, and if they do not apply, the call is redirected
to 'Default Destination' (Closed dates)
• Custom Destinations: Redirects all calls received during set hours to PBXware
extension provided here
• Closed dates: Sets the specific date when all calls are redirected to 'Default
Destination'. If 'Destination' field in the Closed dates is set, call will not go to
'Default Destination' but to this number.
Ringing Type
Ringing Type options
134
• Ringing
Select the ringing type played back to calling party before they enter the IVR.
(ex. Rather than just falling into the IVR, play the ring sound to user or music
files located under the MOH class).
(Select box)
• Music on Hold
Select the MOH/Music on Hold class, played to users after they make a
selection in IVR.
(ex. MOH class usually contains one or more sound files. To see these files go to
'System: MOH').
(Select box)
• Custom ring tone
Ring with custom ringtone phone which is set as a destination in IVR.

(ex. If caller in IVR presses '3' and on that destination there is a Linksys phone,
it will ring with this ring tone, ie. <Simple-2>).
([0-9][a-z])
Local Dialing
IVR Local Dialing
• Dial local/network destinations
IVR option can dial local network or proper/mobile phone numbers. By setting
this option to 'Yes', the IVR will be allowed to dial local network extensions
only.
(ex. If IVR has this options set to 'Yes', only local network extensions will be
accessible from this IVR. No proper/mobile numbers would be dialed).
(Option buttons)
• FAX dialing
Set this option to Yes if you need to send Fax to email of any local extension
which is dialed.
(ex. If John dials into an IVR which has this option turned on, he will have the
ability to dial local extensions number to which system will generate a FAX
tone. He will then send a fax which will be sent to email of the extension that he
dialed).
(Option buttons)
135
• Enable range limit
Tells the system which extensions on the local network IVR will be able to dial.
(ex. Yes, No, N/A)
(Option buttons)
• Start at
Starting number in the available range of extensions.

(ex. 1000)
([0-9])
• End at
End of available range

(ex. 2000)
Permissions
Permissions are used to allow an organization to restrict who is able to enter an IVR. In
particular there are organizations where access to the IVR is only allowed to callers with
a valid account number, but it can be used for other similar purposes.
IVR Permissions settings
• Account Access Only
Allow only certain PBXware extensions to access the IVR.

(ex. If this option is set to 'Yes', only extension numbers set under the 'Account
list' will be allowed to access this IVR).
(Option buttons)
• Account List
Only local extension numbers provided here, separated with a single space, will
be allowed to enter this IVR.
(ex. John dials this IVR, but his extension number is not in the 'Account list'.
John is transferred to 'Sales extension').
([0-9])
• Sales Extension
If caller extension is not provided in the 'Account list' his call will not enter the
IVR, but will be redirected to extension number provided here.
136
(ex. John dials this IVR, but his extension number is not in the 'Account list'.
John is transferred to extension number provided here).
([0-9])
Multi-digit IVR codes

The only difference between standard and multi-digit IVRs is that the latter accepts two
or more digits as a response (numbers between 10-100 for example), therefore
providing a wider range of options.
IVR Access Codes
This screen lists all system Multi-digit DIDs with the following details:
• Access Code
Multi-digit IVR access code.

(ex. 22)
(Display)
• Description
Access code description.

(ex. desc goes here)
(Display)
• Extension
Extension where call will be transferred to.

(ex. 1004)
(Display)
• Edits the IVR configuration
(ex. Click to edit IVR configuration).

(Buttom)
• Deletes an IVR from the system
(ex. Click to delete an IVR from the system).

(Button)
Search
137
Search Access Codes
• Search
Search phrase
([a-z][0-9])
• Access Code
Should search filter be applied to access codes

(ex. Check the box to search access codes).
(Check box)
Add/Edit Access Code
Add/Edit Access Code
• Access code
Option dialed from the Multi-digit IVRs that will transfer user to PBXware
'Extension' number.
(ex. John enters the Multi-digit IVR and dials this 'Access Code' (e.g. 99). When
John dials 99, he will be transferred to PBXware 'Extension' number).
([0-9])
• Description
A short Description of this access code and its destination. This information is
viewed only by the Administrator.
(ex. Once the administrators goes to the 'PBXware: Multi-digit IVR codes', a list
of access codes, their descriptions and destination extensions will be displayed).
([0-9])
• Extension
PBXware extension to which a call will be transferred to once the 'Access code'
is dialed from the Multi-digit IVR.
138
(ex. John enters the multi-digit IVR and dials 99, which is set to lead to
PBXware extension 1000. When John dials 99, he will be transferred to
PBXware 'Extension' provided here).
([0-9])
PIN-based IVR
PIN-based IVR allows dialing local/remote destinations by providing a pre-set IVR PIN
number.
PIN-based IVR window
• PIN
IVR PIN number

(ex. 55555)
(Display)
• Description:
IVR PIN short description

(ex. Remote Cell)
(Display)
• Expiry date:
PIN expiry date

(ex. 31-01-2007)
(Display)
• Destination:
Destination dialed once the IVR PIN is provided

(ex. 061505139)
(Display)
• Edits the IVR PIN configuration
(ex. Click to edit the IVR PIN configuration)

(Button)
• Deletes the IVR PIN settings from the system
139
(ex. Click to delete the IVR PIN settings from the system)
(Button)
TIP:
If the item is grayed out and the 'Expiry date' field underlined, that PIN has expired.
Search
done by PIN numbers
Search PIN
• Search:
Search phrase
([a-z][0-9])
• PIN:
Should search filter be applied to PIN numbers

(ex. Check the box to search PIN numbers)
(Check box)
Upload
Your may upload a .csv file with multiple PIN and Destination codes in one step. Click
the 'Browse' button and select the .csv file on your computer. Then just click the
'Upload' button to add new PIN codes to PBXware.
Upload CSV
Sample .csv file:
10002,John Smith Cell,31-01-2007,061555109
20002,George Nimara Home,31-01-2007,032246509
140
Add/Edit PIN
Add/Edit PIN
To manually add an IVR PIN code, click the 'Add PIN' button on top. The following
options will be displayed.
• PIN:
Unique IVR PIN number. This PIN number is provided once requested by IVR.
A correctly supplied PIN will make PBXware dial the 'Destination' number.
(ex. John dials local IVR (1003) and is asked to provide a PIN. He enters PIN
(55555) and PBXware dials the number provided under the 'Destination' field).
([0-9])
• Description:
Short description of the IVR PIN. This description is used for describing the
Destination number
(ex. 'John's cell phone', or '21255510205')
([a-z] [0-9])
• Expiry date: All PIN codes can be valid until a certain date. You are strongly
encouraged to set the expiration date by clicking on the 'Calendar' icon, next to
the 'Expiry date' field and selecting a desired date. The date can be provided
manually in the following form 'dd-mm-yyyy'. Accessing the PIN after the
'Expiry date' will be impossible.
(ex. If this field is set to expire on 31-01-2007. All IVR calls made with this PIN
by this date will be passed through the Destination number. Calls after the
expiration date will not be made).
([0-9])
• Destination: Ten digit destination number dialed once the IVR PIN is provided.
(ex. Provide a 10 digit number here e.g. 1555102057. Once the IVR PIN is
provided, a call will be made to it).
([0-9])
IVR Tree
141
IVR Tree is a special form of IVRs where the creation of particular IVR is more
graphically oriented than it is with regular IVRs.
IVR Tree
Add/Edit IVR
When you click on the 'Create' button, you will be asked to enter an IVR name and
number after which the IVR will be available for editing.
After you've created or selected the existing IVR, you can edit it
Add/Edit IVR
If you click on the 'Edit' button, you will get the 'Edit Properties' dialog where you can
edit this IVRs properties
IVR Tree Actions
• Name
IVR's name
(ex. Test Tree)
([0-9][a-z])
• Extension
142
IVR's extension number
(ex. 777)
([0-9])
• Number of rings
Number of rings before the caller enters the IVR

(ex. 4)
([0-9])
• Response Timeout
Time period in seconds during which an IVR option must be dialed by the user.
Useful when Local dialing is turned on.
(ex. 4)
([0-9])
• Digit Timeout
Timeout in seconds during which a new digit must be dialed. Useful when Local
dialing is turned on.
(ex. 1)
([0-9])
• Selection Timeout
Timeout in seconds during which the selection must be made.

(ex. 3)
([0-9])
• Direct access
This option, if turned on, disables direct access to this IVR. This means that this
IVR can be accessed only through other IVR.
(Check box)
• Local dialling
Enable user to dial local extensions while inside IVR.

(Check box)
Button 'Delete' of course is used to delete given IVR.
'Save' button must be pressed every time a change has been made so it could take place.
Actions & Selections
143
As you can see in the example picture, a main part of the IVR Tree is Actions and
Selections. Action or Selection can be added by clicking on a plus icon, and can also be
deleted by clicking on the delete button right next to it.
IVR Tree Actions & Selections
Actions are items that are going to be executed when the user enters IVR. They will get
executed in the order in which they are placed in the list. Actions can be added by
clicking on the plus button.
IVR Tree Actions
• Dial Local Extension
This option lets you dial the local extension which is set in the Extension field,
when the action is executed.
(ex. 303)
([0-9])
• Dial Remote Extension
This option lets you dial the remote extension which is set in the Destination
field.
(ex. 404)
([0-9])
• Dial Voicemail
144
Enables you to leave a voicemail for the given extension
(ex. 505)
([0-9])
• Check Voicemail
This option enables you to check the voicemail of the extension which you
provide with your keypad.
• Go to context
Go to given context and extension

(ex. Context: time, Extension; 222)
• Start Recording
Start recording further conversation

(ex. If Silent recording is turned on, the user will not hear the announcement that
his call is recorded)
• Set Queue Priority
If the user is about to enter a queue he is going to enter it with a given priority.
(ex. 5)
(Select box)
• Set Language
Set the language that is going to be used in IVR

(ex. Use the two letter notation like 'us')
([a-z])
• Play Sound
Play a sound file which the user can select and also set a language for it if the
sound file was supplied in several languages. Also you can set here to accept
DTMF tones or keypad buttons during play so you can dial local extensions if
that option is enabled in IVR properties.
• Change CallerID
Change users CallerID

(ex. John Doe)
([0-9][a-z])
• Wait
Wait the given number of second before IVR continues to execute actions.
(ex. 5)
145
([0-9])
• Busy signal
Play the 'Busy' sound to user.
• Hangup
Hang up the current call.
Arrows pointing up or down are used to change the order of items which is very
important for Actions but non-important for Selections. Clicking on the up arrow on an
action puts that action in front of the previous item, or clicking on the down arrow puts
that action after the next item.
IVR Tree Graph

When you click on the IVR Tree Graph menu, you will get a graphical representation of
your IVR Tree
IVR Tree Graph
146
MT 3.8.5 Queues
PBXware Queue system allows you to receive more calls in your PBXware than your
staff members are able to answer at the same time.
Queues
They will enable you to deal with your calls more efficiently and your customers will be
held in a queue, listening music on hold and receiving information messages about
aproximate wait time and/or their position in queue.
Queues consist of:
• Incoming calls being placed in the queue

• Members that answer the queue (extensions or users that login as agents)
• A strategy for how to handle the queue and divide calls between members
• Music played while waiting in the queue
• Announcements for members and callers
In Multi Tenant PBXware, only available Queue members are SIP Members, extensions
added to the queue.
147
Contents
• 1 Queues
• 2 Search
• 3 Add/Edit Queue
o 3.1 Queue members and agents
• 4 Advanced Options
o 4.1 General
o 4.3 Redirect Extension
o 4.4 Queue Timers
o 4.5 Empty Queue
o 4.6 Greeting
o 4.7 Position Announcements
o 4.8 Agents
o 4.9 Agents Announce
o 4.10 Recordings
o 4.11 Incoming Options
o 4.12 Exit Digit
148
Queues
This screen lists all the system queues with the following details:
Queues
• Name:
Queue name
(ex. Patience)
(Display)
• Number:
Queue network number

(ex. 1001)
(Display)
• Edit
Edits the queue configuration

(ex. Click to edit the queue configuration)
(Button)
• Delete
Deletes a queue from the system

(ex. Click to delete a queue from the system)
(Button)
Search
By selecting the 'Search' Command, a search menu will be displayed. Searches can
be done by Name and Number
149
Search
• Search:
Search phrase
([a-z][0-9])
• Name:
Should the search filter be applied to queue names

(ex. Check the box to search queue names)
(Check box)
• Number:
Should the search filter be applied to queue numbers

(ex. Check the box to search queue numbers)
(Check box)
Add/Edit Queue
Clicking on 'Add/Edit' Queue will open standard options shown below
Add/Edit Queue
• Queue Name:
Unique queue network name/identifier

(ex. Provide a unique queue identifier/name here)
([a-z][0-9])
• Queue Number:
Unique network queue number
150
(ex. This number is to be dialed in order to access the queue)
([0-9])
• Max Callers:
Maximum number of callers allowed to wait in a queue at the same time. This
number should be set in accordance with the number of members answering the
queue calls
(ex. If this field is set to 4, only 4 callers will be allowed to enter the queue. If caller
number 5 tries to enter the queue he will be transferred to <%PRODUCT%> 'Redirect
Extension' number).
([0-9])
• Queue members and agents:
This button opens up a screen in which you can add and manage queue members
(ex. Click on the button to manage queue members)
(Button)
Queue members and agents
When you click on the Queue members and agents button, you will get the following
window where you can manage members of the current queue.
Queue members and agents
This window is divided into two tables:
• Available - which is positioned on the left and shows all the extensions on the
system not assigned to this queue.
• Members - which is located on the right and shows extensions assigned to this
queue.
151
Also, every table has the search ability so you can search for members names,
numbers and type.
• #
Checkbox which is used to select extensions for adding or removal from queue
(ex. Clicking on any checkbox next to an extension will show arrow buttons for
adding or removal in the middle which depends on in which table checkbox is
selected).
(Check box)
• Name
Extension name
(ex. Sales 1)
([0-9][a-z])
• Number
Number of the extension on the system

(ex. 1001)
([0-9])
• Type
Option used to filter the list by the member type.

Any - displays all entries
SIP - displays only SIP static members (SIP extensions)
IAX - displays only IAX static members (IAX extensions)
• Penalty
This value can be set per member, so the system can decide which priority to use
when sending calls to members
(ex. Set values between 0 and 10 where 0 is the highest priority and 10 is the lowest
priority. This works best when the ring strategy is ringall).
([0-10])
Advanced Options
General
152
Advanced Options
• Music On Hold:
Select MOH(Music On Hold) class name. All sound files belonging to this MOH class
will be played to users in queue
(ex. User A enters the queue. After the greeting message is heard, all sound files
belonging to selected MOH class are played in the background)
(Select box)
• Rings to Answer
How many rings will be played to a caller before call enters the queue.
Set this value to 0 if you would like calls to hear Queue music on hold as soon as call
is connected.
• RTP Delay [sec]:
Delay time in seconds inserted before the queue greeting message is played. This
solves the 'half-played' file problem. Keep this value between 1-3
(ex. User A enters the queue and hears '..r call is first in line...' and wonders what
was that.. what did the voice say!? Set this field to 1 so that 1 second pause is added
before the message is played. Now, when user A enters the queue he will hear 'Your
call is first in line...').
([0-9])
• Replace Caller ID:
Replace the caller id with the custom value

(ex. Type 'Lobby - %CALLERID%' to display the caller id information as 'Lobby -
5552879' where 5552879 is the actual number calling in)
([a-z][0-9])
153
• Language:
Define custom language for all sound files played by the queue
(ex. To play Spanish sound files to all users waiting in the queue, type 'es' here.
NOTE: PBXware MT comes with English sound files by default. To install sound files
in other languages, please see ' Settings: Protocols: Sip: Language' for more
information)
([a-z])
• Queue URL:
Send the following URL to agents softphone, if it supports that option, and the
softphone can open that URL so agent can see callers additional info if available. This
option must be supported by softphone for it to be used.
(ex. Additional info on Queue URL field can be obtained here)
([a-z][0-9])
• Custom ring tone:
Ring agent phones with custom ringtone set here

(ex. <Simple-2>)
([0-9][a-z])
• Custom sounds folder:
When Queue Callback is turned on, you will be able to add custom queue sounds
that will be played in that case.
Example:
Create a new folder for these sounds in
/opt/pbxare/pw/var/lib/asterisk/sounds/
and put your custom queue sounds here.
([a-z][A-Z])
TIP: Sounds should be named as follows:
• queue-youarenext - Your call is first in line and will be answered by the next
available representative)
• queue-thereare - You are currently caller number
• queue-callswaiting - Waiting to speak with a representative
• queue-holdtime - The estimated call time is currently
• queue-minutes - Minutes
• queue-seconds - Seconds
• queue-lessthan - Less than
154
• queue-thankyou - Thank you for your patience
• queue-periodic-announce - All of our representatives are currently busy.
Please stay on the line and your call will be answered by the next available
representative
Operation Times
Set the queues open/closed times. Depending on the time when call is received, the
call can be redirected to different PBXware MT destinations
Operation Times

(ex. Yes, No)
(Option buttons)
PBXware MT extension all calls are redirected to during the closed time hours
(ex. 1000)
([0-9])
• Greeting:

(ex. greeting-***)
(Select box)
Description of destinations follows in this priority order:
• Open dates: Sets the working hours during which DID is to redirect calls as
set in DID Add/Edit window. If any call is received during the hours not set
here, 'Custom Destination' are checked, and if they do not apply, the call is
redirected to 'Default Destination' (Closed dates)
155
• Custom Destinations: Redirects all calls received during set hours to the
PBXware MT extension provided here
• Closed dates: Sets the specific date when all calls are redirected to a 'Default
Destination'. If 'Destination' field in the Closed dates is set, call will not go to
'Default Destination' but to this number.
Redirect Extension
Redirect Extension
• Redirect Extension:
Max callers redirect extension number

(ex. If queue 2000 accepts a maximum of 4 users waiting at the same time, any new
user that enters queue 2000 will be redirected to the extension number provided
here)
([0-9])
• Redirect to Voice Mail:
Choose whether the redirect extension is a voicemail

(ex. Yes, No, N/A)
(Option buttons)
Queue Timers
• Queue Ring Timeout:
Number of seconds during which the members will be rung, after which the Position
Announcement will be played to the caller, if it is enabled
(ex. Members extensions are ringing for 10 seconds. After that time, Position
Announcement is played to caller and then Agent(s) are rung again).
([0-9])
• Prioritize Queue Timeout
Used to control the priority of the two possible timeout options specified for a
queue. Queue Ring Timeout field has a timeout value that can be specified to control
156
the absolute time a caller can be in the queue. The timeout value in queues.conf
controls the amount of time (along with retry) to ring a member for. Sometime these
values conflict, so you can control which value takes precedence. The default N/A
setting will use Queue Ring Timeout value, same as setting this to Yes. No will
override Queue Ring Timeout field and use value from queues.conf.
Queue Timers
• Max Wait Seconds:
Maximum time a caller can wait in a queue. Once this time is exceeded caller will be
redirected to 'Max Wait Extension' number.
(ex. User A is waiting 5 minutes already in the queue. If this field is set to 300(300s =
5min), A will be redirected to 'Max Wait Extension' so he doesn't lose his mind
waiting)
([0-9])
• Max Wait Extension:
This option works along with the 'Max Wait Seconds' field. Provide the extension to
which caller will be redirected once time set under 'Max Wait Seconds' exceeds.
(ex. User A is waiting 5 minutes already in the queue. If 'Max Wait Seconds' field is
set to 300(300s = 5min), A will be redirected to this extension so he doesn't lose his
mind waiting)
([0-9])
Empty Queue
• Join Empty
Set whether a caller can join a queue if no agent is logged in or unavailable

Example:
It is recommended to set this option to 'No'. Do not allow user to enter the queue if
the call will not be answered by anyone. Following options are available:
157
• Yes - Join queue if no agents or only unavailable agents are in the queue
• No - Do not join queue if no agents available
• Strict - Do not join queue in no agents or only unavailable agents are in the
queue
(Option buttons)
• Leave when empty
When there are no agents inside a queue, callers will leave it depending on the
setting
Example:
• No - This option is disabled

• Yes - If you wish to remove callers from the queue if there are no agents
present
• Strict - If there are calls queued, and the last agent logs out, the remaining
incoming callers will immediately be removed from the queue
(Select box)
• Empty Destination:
When queue doesn't have any agent logged in, redirect callers to following
destination
(ex. 7001)
([0-9])
Greeting
Greeting
• Greeting:
Select a greeting file to play to all callers waiting in a queue.

(ex. Record a custom sound greeting file (e.g. 'All our representatives are busy...')
and select that file. This file will be played to all callers once they enter the queue).
(Select box)
NOTE: File name has to be in the following format 'queue-greeting-$NAME.gsm'
• Wait Seconds:
158
Delay time in seconds inserted before playing the greeting message. This delay is
useful when users do not hear the beginning of the greeting message. Keep this
value between 1-3 seconds.
(ex. John enters the sales queue, but cannot hear the beginning of a greeting sound
file file (e.g. ..ur representatives are busy...). Set this option to 1. This will insert one
second of silence before the greeting file is played and should fix the partial sound
file error. Now, all callers entering the queue should hear the full greeting message
(e.g. All our representatives are busy...))
([0-9])
Position Announcements
Position Announcements
There are two types of position announcements: 'Hold Time' (Tells the queue
position) and 'Periodic Announcements' (Plays custom message)
• Announce Hold-Time:
Enable callers waiting in a queue to hear the hold-time announcements.

(ex. Setting this option to 'Yes' will enable the hold-time announcements. A single
caller waiting in a queue would hear a 'Your call is now first in line and will be
answered by the next available representative. Thank you for your patience'
message).
(Option buttons)
• Announce Frequency:
How often to play the hold-time announcement message (time in seconds).

(ex. If this field is set to 30, a single caller waiting in queue will hear 'Your call is now
first in line and will be answered by the next available representative. Thank you for
your patience' message every 30 seconds).
([0-9])
NOTE: If you set this option to '0', the announce message will not be played.
159
• Min. Announce Frequency
The minimum number of seconds between position announcements. Default value is

15 seconds.
• Announce Round Seconds:
This feature rounds announcement minutes and seconds to specific format.

([0-9])
• Periodic Announce:
Select the sound file that is played periodically to callers waiting in a queue.
(ex. Record a message 'Hang in there buddy!' and set it as a periodic announcement.
This message will be played to callers every 'Periodic Announce Frequency' seconds)
(Select box)
NOTE: File name has to be in the following format 'periodic-announce-$NAME.gsm'
• Periodic Announce Frequency:
Time interval in seconds at which the periodic message is to be played.

(ex. If this field is set to 30, all callers waiting in queue will hear the 'Periodic
Announce' message every 30 seconds, as long as they stay in the queue).
([0-9])
• First Periodic Announce Delay
The minimum number of seconds between position announcements. Default value is

15 seconds.
Agents
Agents
160
• Send Manager events
Options: Yes, No
When this option is set to yes, the following manager events will be generated:
AgentCalled, AgentDump, AgentConnect and AgentComplete.
Ring Strategy
Set the way calls are transferred to queue members answering the calls.
Available options:
• ringall - Ring all available Members until one answers (default)

• linear - When used with static members linear ringing strategy will ring
interfaces in the order specified in queues.conf configuration file for that
specific queue, always starting at the beginning of the list. If used with
dynamic members, the members will be rung in the order in which they were
logged in.
• leastrecent - Ring Member with the least Queue calls
• fewestcalls - Ring Member with the fewest completed Queue calls
• random - Ring random Member
• rrmemory - Round robin with memory. Remember where we left off
• rrordered - Same as rrmemory, except the queue member order from config
file is preserved so interfaces will ring in the order specified in this
configuration file. If you use dynamic members, the members will be rung in
the order in which they were added.
• wrandom - Rings random interface, but uses the member's penalty as a
weight when calculating their metric. So a member with penalty 0 will have a
metric somewhere between 0 and 1000, and a member with penalty 1 will
have a metric between 0 and 2000, and a member with penalty 2 will have a
metric between 0 and 3000. Please note, if using this strategy, the member
penalty is not the same as when using other queue strategies. It is ONLY
used as a weight for calculating metric.
(Select box)
• Autofill:
Should callers be served one by one or in parallel fashion

(ex. With this option turned 'Off', even if there are five agents available calls will not
be transferred to them until first caller waiting in a queue is connected to an agent.
When first caller gets served, caller number two gets served and all other keep
waiting. Obviously it is recommended to keep this feature always turned 'On' so
callers can be served in parallel)
(Option buttons)
• Ring Agents in Use:
161
Should agents in use be rang when new caller comes into queue
(ex. If agent is already in active conversation, with this option set to 'Yes' Agent
extension will ring when new caller enters the queue)
(Option buttons)
• Auto Pause
Autopause will pause a queue member if they fail to answer a call.

Options: No - Member will not be paused, Yes - Member will be paused only in the
queue where the timeout took place, All - Member will be paused in all queues
he/she is a member.
• Member Delay:
This field is the same as RTP Delay, only this option is set for agent answering the
queue calls. Before the call is transferred, custom queue information can be played
to an agent so that agent knows from which queue the call is coming from. This
solves the 'half-played' file problem. Keep this value between 1-3
(ex. Agent X is to answer the call coming from the queue. If the 'Queue Announce' is
set to play custom sound file('This call comes from the Lobby Queue') but only '...s
call comes...' is heard, set this field to 1 so that 1 second pause is added before the
message is played and entire message is played 'This call comes from...').
([0-9])
• Retry All Timeout
Time interval in seconds, for how long to wait before trying the queue agent again.
(ex. If agent Smith cannot answer the incoming queue call and hangs up the ringing
line, the call will not be transferred back to his extension for a time in seconds
provided in this field).
([0-9])
• Timeout Restart:
Reset the internal timer if BUSY or CONGESTION is received from agent

(ex. A call enters the queue and is transferred to Agent X. Agent X has a Polycom
phone (for example) and sends a BUSY signal by hitting the 'Reject' key. This will
reset the internal timer. The call will be transferred to other queue agent(s). The
original agent will not be contacted until all other agents are tried and the call does
not get answered by any of them).
(Option buttons)
• Wrap-up time
162
After a successful call, this will determine how long to wait (time interval in seconds)
before sending a new call to a potentially free agent/member.
(ex. 4)
([0-9])
Agents Announce
Agents Announcements
• Agent Announce
Select pre-recorded message that will be played to the agent before the call is
connected.
NOTE: In order to be displayed in the drop down list, sound file must have 'agent-
announce-' prefix.
• Report Holdtime
Set this option to yes to inform the agent for how long caller was in queue before
the call was connected.
Recordings
Recording
• Record Queue Calls:
Once this feature is activated, all queue calls will be recorded in the desired sound
format.
(ex. John enters the 'Sales' queue and is transferred to 'Queue Agent Smith'. Their
entire conversation is recorded and available for review from 'Reports').
(Option buttons)
• Use MixMonitor:
163
MixMonitor allows you to record conversations with the possibility to adjust the
heard and spoken volume and to append the next conversation in the same file. So,
at the end of the day, you could have all the conversations on one channel in one
file. They will be stored in the same sequence, as they are made.
(ex. Yes, No, N/A)
(Option buttons)
• Monitor format:
Select the audio format all queue calls with be recorded in. Available formats: gsm,
wav and wav49.
(ex. John enters the 'Sales' queue and is transferred to 'Queue Agent Smith'. Their
entire conversation is recorded and available for review from 'Reports').
(Option buttons)
Incoming Options
Incoming Options
• Incoming Options Set the advanced queue call options.
Available options:
• t - allow the called user to transfer the calling user

• T - allow the calling user to transfer the call
• H - allow the caller to hang up by hitting *
• n - no retries on the timeout; will exit queues and go to the next step
• r - ring instead of playing MOH
([a-z])
• Ring ('r') timeout
When 'r' is selected as Incoming Options and this timeout is used, queue will first
ring entered number of seconds after which MOH will start
(ex. 10)
([0-9])
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Dial Options:

channel(s) until one answers. Use with care and don't insert this by default into all of
• R - Indicate ringing to the calling party when the called party indicates ringing, pass
no audio until answered. This is available only if you are using kapejod's bristuff.
• m - Provide Music on Hold to the calling party until the called channel answers. This
is mutually exclusive with option 'r', obviously. Use m(class) to specify a class for the
music on hold.
• o - Restore the Asterisk v1.0 Caller ID behavior (send the original caller's ID) in
Asterisk v1.2 (default: send this extension's number)
• M (x) - Executes the macro (x) upon connect of the call (i.e. when the called party
answers)
• h - Allow the called party to hang up by dialing *
command
• P (x) - Use the Privacy Manager, using x as the database (x is optional)
context.
• G (contextêxten^pri) - If the call is answered, transfer both parties to the specified
• A (x) - Play an announcement (x.gsm) to the called party.
• S (n) - Hang up the call n seconds AFTER the called party picks up.
• d: - This flag trumps the 'H' flag and intercepts any dtmf while waiting for the call to
be answered and returns that value on the spot. This allows you to dial a 1-digit exit
• D (digits) - After the called party answers, send digits as a DTMF stream, then
connect the call to the originating channel.
• L (x[:y][:z]) - Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z'
ms) Only 'x' is required, 'y' and 'z' are optional. The following special variables are
optional for limit calls: (pasted from app_dial.c)
o + LIMIT_PLAYAUDIO_CALLER - yes|no (default yes) - Play sounds to the
caller.
o + LIMIT_PLAYAUDIO_CALLEE - yes|no - Play sounds to the called party.
LIMIT_WARNING_FILE is not defined, then the default behavior is to
announce ('You have [XX minutes] YY seconds').
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• f - forces callerid to be set as the extension of the line making/redirecting the
outgoing call. For example, some PSTNs don't allow Caller IDs from other extensions
than the ones that are assigned to you.
Exit Digit
Exit Digit
• Use Exit Digit
Should users be able to exit the queue by dialing a single digit and be automatically
redirected to a preset PBXware destination.
Example:
John enters the sales queue. A greeting message explains that the user may quit the
queue by pressing the digit defined under the 'Exit Digit' option, and be transferred
directly to the operator or some other destination (set under 'Extension' option).
(Options buttons)
• Exit Digit
Define the exit digit here. Once this digit is dialed by the user waiting in queue, it will
transfer the call directly to 'Extension' destination
Example:
John enters the sales queue. The greeting message explains thatthe user may quit
the queue by pressing the digit defined here (e.g. 9) and is transferred directly to the
operator or some other destination (set under 'Extension' option).
([0-9])
• Extension
Local PBXware extension number that is dialed once the 'Exit Digit' is dialed.
166
(ex. John enters the sales queue. The greeting message explains that the user may
quit the queue by pressing the digit defined under the 'Exit Digit' option, and be
transferred directly to the PBXware extension defined here).
([0-9])
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MT 3.8.5 Voicemail
PBXware voicemail is an advanced answering machine. Although each extension is
equipped with a voice mailbox, voice mailboxes can be created on its own as well from
this location.
Voicemail
Contents
• 1 Search
• 2 Mailboxes
o 2.1 Add/Edit Voicemail
• 3 Groups
o 3.1 Add/Edit Group
Search
By selecting 'Search' Command, search menu will be displayed. Searches can be done
by Name, Email and Extension number
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Search
• Search
Search phrase
(ex. Depending on which check boxes are selected below (Name, E-mail, Number)
provide the corresponding phrase here. For example, if the e-mail is selected below,
type some email address here e.g. [email protected] and click the search icon or hit
enter on the keyboard)
([a-z][0-9])
• Name
Search voice mailboxes by user name

(ex. Check this box and under 'Search', type the user's name or surname and click the
Search icon or hit enter on the keyboard to display results)
(Check box)
• E-mail
Search voice mailboxes by email address

(ex. Check this box and under 'Search', type the user's email address and click the
Search icon or hit enter on the keyboard to display results)
(Check box)
• Extension
Search voice mailboxes by network extension

(ex. Check this box and under 'Search', type the voice mailbox network extension and
click the search icon or hit enter on the keyboard to display results)
(Check box)
Mailboxes
This screen lists all system mailboxes with the following details:
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Mailboxes
• Name:
Full name of the voice mailbox user

(ex. Peter Doyle)
(Display)
• Mailbox:
Voice mailbox extension number

(ex. 1006)
(Display)
• Domain:
Domain/Context voice to which the mailbox belongs

(ex. t-200 for Tenant #200)
(Display)
• Edits the voice mailbox configuration
(Click to edit the voice mailbox configuration)

(Button)
• Deletes a voice mailbox account from the system
(ex. Click to delete a voice mailbox from the system)

(Button)
TIP
Edit and Delete commands will be disabled for users with the system extension. Their
voicemail settings are edited via self care or by editing their extension.
Voice Mailbox Access:
In order to access custom voice mailboxes from any PBXware extension, dial *124 ,
after you hear message "Comedian mail mailbox" enter
$VOICE_MAILBOX_NUMBER', you will be prompted to enter PIN for that extension.
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Example: From extension 1001 dial *124 and after "Comedian mail mailbox" message
is played enter 2000 to get to the mailbox for extension 2000, After you enter correct
PIN you will be granted with access to the voicemail box for extension 2000.
Add/Edit Voicemail
Clicking on 'Add Voicemail/Edit' Voicemail will open the voicemail screen shown
below.
Add/Edit Voicemail
• Mailbox:
Unique network voice mailbox extension number

(ex. Set this field to 5001, for example. Now, in order to dial into this voice mailbox,
simply dial 5001 from any PBXware extension)
([0-9])
• Name:
Full name of the voice mailbox owner

(ex. John Smith)
([a-z])
• PIN (Personal Identification Number):
Four digit number used for voice mailbox authentication
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(ex. Each voice mailbox has a unique PIN. In order to login to your voice mailbox,
provide this number when asked for it by the operator e.g. 1947)
([0-9])
• Email:
Email address associated with the voice inbox. This email is used for new voice
message notification and audio file attachments
(ex. If '[email protected]' is set here, once this mailbox receives a new message,
notification and attached voice message (depending on if this option is enabled) is sent
to this email address)
([a-z][0-9][@_-.)
• Language:
If you installed additional language sound files to the system you can set different
language to be used for sound files related to voicemail
(ex. If you installed French language files under sounds/fr folder (as per online how to)
put 'fr' here to play French sound files when answering the voicemail)
([a-z])
• Send E-mail:
Whether or not to send an e-mail to the address given above

(Yes, No, N/A)
(Option buttons)
• Pager e-mail:
Provide the pager e-mail address here

(ex. If '[email protected]' is set here, once this mailbox receives a new message,
notification is sent to this pager email address)
([a-z][0-9][@_-.])
Greeting message played to uses before they are transferred to voice mailbox to leave
the message
(ex. Mailbox user may choose between a 'Busy' and an 'Unavailable' message)
(Select box)
• Unavailable message:
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Upload the custom unavailable message. Unavailable message supports: WAV, wav,
and gsm files only
(ex. If the default unavailable message does not suit your needs, click the 'Browse'
button, upload a custom message, and select it under 'greeting message')
(Select box)
• Reset Unavailable message:
Reset the current unavailable message

(ex. To reset the current unavailable message, click this button. The message will be
deleted from the filesystem)
(Option buttons)
• Busy message:
Upload the custom busy message. The busy message supports: WAV, wav, and gsm
files only
(ex. If the default busy message does not suit your needs, click the 'Browser' button,
upload a custom message, and select it under 'greeting message')
(Select box)
• Reset Busy message:
Reset current busy message

(ex. To reset the current busy message, click this button. The message will be deleted
from the filesystem)
(Option buttons)
• Skip instructions:
Skip the instructions telling users how to leave a voice message

(ex. Once the caller reaches the voice mailbox, instructions on how to leave voice
message are played. You are encouraged to set this option to 'Yes' all the time)
(Option buttons)
• Attach:
Should the voice message be attached and sent along with the notification email
(ex. Caller leaves a voice message to John. With this option set to 'Yes', notification
email John gets will have a voice message attached to it so John can listen to it without
signing in to his voice mailbox)
(Option butons)
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Should the voice message sound file be deleted from the filesystem after sending it as
an attachment to the user's email address
(ex. Caller leaves a voice message to John. With this option set to 'Yes', the voice
message will be deleted after sending it as an attachment to John's email address)
(Option buttons)
• Say CallerID:
Should the extension number which left the voice message be announced to the
mailbox owner
(ex. With this option set to 'Yes', John will hear '... from phone number 1004...' when
checking mailbox, for example).
(Option buttons)
Allow the user to review his voice message before committing it permanently to the
voice mailbox
(ex. After a caller leaves a voice message and presses '#', additional review options are
allowed: 1 to accept the recording, 2 to re-record your message, etc.
(Option buttons)
• Allow Operator:
Allow the caller to reach the operator from the voice inbox by pressing '0'
(ex. Once the user leaves a voice message and presses #, additional options, including
'...press 0 to reach an Operator' are heard)
(Option buttons)
• Operator Extension:
The local extension number that is dialed once '0' is pressed to reach the Operator
(ex. Once the caller leaves a voice message to John and presses '0' to reach the
Operator, the extension number provided here (e.g. 1001) will be dialed)
([0-9])
• Play Envelope Message:
Announces the date and time when the voice message was left in the inbox
(ex. With this option enabled, John will hear 'First message, 11:52, 02 Feb 2007' for
example, when checking his voice mailbox)
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([0-9])
This option will allow you to hide your voicemail extension from the directory list.
(ex. Yes, No, N/A)
(Option buttons)
Number of rings played to the caller before the call is allowed to enter Voicemail
(ex. Rather than just 'falling' into Voicemail, it is recommended to set the number of
ring sounds played to caller)
NOTE: By default, this field is empty which means that there isn't going to be any
ringing. Caller will 'fall' into Voicemail
([0-9])
Delay time in seconds inserted before the Busy/Unavailable message is played to

caller. This solves the 'half-played' file problem. Keep this value between 1-3
(ex. Caller is to leave a voice message to John. It hears '...ot at home right now...'.
Adding '1' to this field will add a one second pause before the message is played. So,
now new callers will hear the greeting message without the first part being cut off 'I
am not at home right now...').
([0-9])
• Timezone:
Set the correct date and time format for message envelope. Timezones are taken from
'/usr/share/zoneinfo' system directory.
(ex. Some countries prefer time format in the mm-dd-yy or dd-mm-yy format. Select
among the available options)
(Select box)
Disk Space Used By Voicemail Recording
With a continuous tone for 60 seconds:
• wav49 = 91.0kb
• wav = 863.0kb
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• gsm = 91.0kb
With a continuous silent tone (without sound) for 60 sec:
• wav49 = 0.38kb
• wav = 3.0kb
• gsm = 0.32k b
Groups
Voicemail groups are used to group voicemail inboxes. Once a voice message is left to
group 2002, for example, all destinations that belong to that voicemail group will
receive the same voice message. This screen lists all system voicemail groups with the
following details:
Groups
• Group:
Voicemail group name/identifier

(ex. Lobby)
(Display)
• Extension:
Voicemail group extension. Once dialed, the voicemail message will be left to all
'Destinations' voice boxes
(ex. 2002)
(Display)
• Destinations:
Voicemail boxes assigned to a voicemail group

(ex. 1001, 1002, 1003)
(Display)
• Edits the voicemail group configuration
(ex. Click to edit the voicemail group configuration)
176
(Button)
• Deletes a voicemail group from the system
(ex. Click to delete a voicemail group from the system)

(Button)
Add/Edit Group
Add/Edit Group
• Name:
Voice mailbox network name

(ex. Create a voicemail group 'Lobby' for example)
([a-z][0-9])
• Extension:
Voice mailbox network extension number

Example:
To leave a message to this group mailbox, callers will have to dial the extension
number set here, e.g. 2002)
([0-9])
• Mailboxes:
A list of mailboxes that belong to this voicemail group is set here

Example:
To assign extensions to this voicemail group, provide their extension numbers

separated by a comma ',' here. For example, 1001,1002,1003...)
177
([0-9])
• Incoming Limit (per call)
Maximum number of times each call can enter the Voicemail group. Entering some
value in this field will prevent eventual infinite loops that might be accidentally set up
on the system.
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MT 3.8.5 Monitor
Contents
• 1 Monitor
o 1.1 Monitoring on master tenant
o 1.2 Monitoring on slave tenant
Monitor
Monitoring window allows the administrator to monitor all PBXware MT extensions,
trunks, conferences, queues and live channels in real time
Monitor
Monitoring on master tenant

When on master tenant, you are able to monitor all the trunks and live channels.
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Trunks
Trunks
Monitored trunks are displayed in real time with the following details:
• Name:
Trunk name
(ex. Depending on the provider settings this can be set to a phone number, ip address
or some context)
(Display)
• IP:
Provider IP address
(ex. 203.196.128.5)
(Display)
• Status:
Displays the trunk status (online/offline)

NOTE: Please set the 'Qualify' = '2500' in the Trunk settings to see its status
(ex. If the 'Qualify' trunk option is empty, 'Unmonitored' is displayed here. Otherwise,
'(e) ok (159ms)' is displayed, for example)
(Display)
Search
Search
• Refresh
Time interval in seconds at which data details should be refreshed

(ex. Select '10 sec' in this field, for example, and click the 'Update' button)
180
(Select box)
• Protocol
Filter the data based on the protocol type (ALL, SIP, IAX)
Example:
Select 'ALL', for example, and click the 'Sort' button to display both SIP and IAX
extensions
(Select box)
• Status:
Sort extensions based on their network status (ALL, Online, Offline)

Example:
Select 'Online', for example, and click the 'Sort' button to display extensions that are
registered/online only
(Select box)
• Letter
Sort extensions based on the user name they belong to

(ex. Select 'B' for example, to display extensions that belong to users whose names
start with letter B (e.g. Brown James))
(Select box)
Live Channels
Live Channels
Monitored live channels are displayed in realtime with the following details:
• From:
SIP/EXTENSION number of user making the call

(ex. SIP/6464)
(Display)
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• To:
Extension number of user receiving a call

(ex. SIP/6464)
(Display)
Refresh
Refresh
• Refresh

Example:
Select '10 sec' in this field, for example, and click the 'Update' button
(Select box)
Actions
Actions
• Hangup:
Hangup active conversation

(ex. Let's say that extension 1000 is talking with queue agent 1001. Select the box
under 1000 and click this button. Conversation between these two extensions will be
terminated).
(Button)
Monitoring on slave tenant

When on slave tenant, you are able to monitor all the extensions from that slave.
182
Monitoring on slave tenant
Monitored extensions are displayed in real time with the following details:
• Name
Name to which the user extension is registered

(ex. Peter Doyle)
(Display)
• Extension
Protocol used by the extension/Extension network number

(ex. SIP/2002)
(Display)
• IP Address
IP address:port from which the UAD/Phone registers

(ex. 192.168.1.1:5060)
(Display)
• Status
UAD/Phone network status (Online/Offline) + (ping time)

(ex. Online (56ms)/Offline)
(Display)
• User Agent
UAD/Phone Brand/Version
(ex. Grandstream 101)
(Display)
• On Call
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Is the user participating in a conversation at this moment
(ex. Yes, No)
(Display)
If you click on an IP address, it will open up a new window showing phones web
interface, if it has one enabled.
TIP: Certain call actions (such as transferring calls, hanging up etc...) can be performed
on active calls as well
Search
Search
• Refresh

Example:
Select '10 sec' in this field, for example, and click the 'Update' button
(Select box)
• Protocol
Filter the data based on the protocol type (ALL, SIP, IAX)
Example:
Select 'ALL', for example, and click the 'Sort' button to display both SIP and IAX
extensions
(Select box)
• Status
Sort extensions based on their network status (ALL, Online, Offline)

Example:
Select 'Online', for example, and click the 'Sort' button to display extensions that are
registered/online only
(Select box)
184
• Letter:
Sort extensions based on the user name they belong to

Example:
Select 'B', for example, and click the 'Sort' button to display extensions that belong to
users whose names start with letter B (e.g. Brown James)
(Select box)
Actions
Actions
• Monitor
Listen to active conversations. Select one of the active conversations under 'Channels',
click this button and provide the extension number that is listening to the active
conversation.
NOTE: You may listen to active conversations by dialing *199 + $EXTENSION number as
well. But no matter which method you listen to the calls with, the listen service has to
be enabled in the enhanced services of the extension that listens the call.
(ex. Let's say that extensions 1000 and 1001 are in conversation. Select this button and
type 1005 into the popup window. Extension 1005 will ring and once the handset is
picked up, active conversation will be heard).
(Button)
• Transfer
Transfer a party from the active conversation to different destination

(ex. Let's say that extensions 1000 and 1001 are in conversation. Select one extension
(e.g. 1000) under 'Channels' and click this button. Type 1005 into popup window.
Extension '1001' will be transferred to extension '1005')
(Button)
• Hangup
Hangup the active conversation

(ex. Let's say that extensions 1000 and 1001 are in conversation. Select one extension
(e.g. 1000) under 'Channels' and click this button. Conversation between these two
extensions will be terminated).
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(Button)
• Reboot
This option is currently used to reboot the Snom phones

(ex. If you have a Snom phone that is online, select 'Reboot' from 'Channels' select box
of that phone, and click on the 'Reboot' button) (Button)
MT 3.8.5 Reports
Contents
• 1 Reports
o 1.1 Reports on master tenant
o 1.2 Reports on slave tenants
Reports
Reports display detail records of all PBXware MT calls, system action logs, CLI
messages and SMTP logs
Reports
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Reports on master tenant
Master tenant has the ability to view reports from all tenants.
CDR
CDR displays detailed records of all PBXware MT calls with the following details
CDR
• Tenant
Tenant on which the call happened.

(ex. If call was made on tenant with prefix 300, 300 will be shown here).
(Display)
• From
Extension number from which the call was made

(ex. If call was made from extension 1001 to extension 1004, '1001' is displayed here).
(Display)
• To
Extension number to which the call was made

(Display)
• Date/Time
Date and Time when the call was made

(ex. 04 Oct 2006 10:44:10)
(Display)
187
• Duration
Call duration time in hh:mm:ss format

(ex. 00:12:45)
(Display)
• Billing
Time billed by the system

(ex. 00:12:45)
(Display)
• Status
Displays the call status

Example:
Depending on whether a call was answered or not, this field value may have the
following content:
• Answered
• Not Answered
• Busy
• Error
(Display)
• This icon is displayed once a call is recorded and 'Delete' or 'Listen' enhanced
service is active
• This is a box used with the CDR commands to select a desired call
Search/Filter
Search/Filter
• Start Date
Select a Search /Filter start date

(ex. Click on the small 'Calendar' icon next to a field and select the desired date)
(Option button)
188
• End Date
Select a Search/Filter end date

(ex. Click on the small 'Calendar' icon next to a field and select desired date)
(Option button)
• From
Select whether you want to search CDRs by Destination(s) or Trunk from where the
call got in
(ex. Destination(s) or Trunks)
(Select box)
• To
This field points to the Destination(s) or Trunk for which you are searching
(ex. Trunk)
(Select box)
• ID
ID of the CDR. When the user selects a CDR, the ID field shows that CDR's ID. If there
are problems on the system, the customer can supply support team with a
problematic CDR ID which helps in locating it.
(ex. 1221447123.66)
([0-9] .)
• Tenants
Type one or multiple tenant numbers (comma separated) to filter records from certain
tenants only.
(ex. 300)
([0-9])
• Start Time
When searching for CDRs, this is the start time in the Start Date
(ex. Time in hh:mm:ss format like 10:15:30)
([0-9] :)
• End Time
When searching for CDRs, this is the end time in the End Date
189
([0-9] :)
• From
If you chose Destination(s) in the From Select box, you will enter an extension from
which the call came. If your selection was Trunk, you will have a Select box in this place
where you can choose a trunk on the system from which the call came.
Example:
• Destination(s) - 1009
• Trunk - Sales
([0-9] or Select box)
• To
Here you will enter number of the destination or select a trunk in which the call ended
up.
(ex. 1007)
([0-9])
• Status:
Search calls by selecting desired call status

Example:
Click on a 'Please Select' button and select one of the available fields:
• All
• Answered
• Not Answered
• Busy
• Error
(Select box)
TIP: After making any changes to search filter, be sure to click the search icon
Actions
Listen
• Listen:
190
Once the 'Listen' icon is displayed next to a call record it means that the specific call
was recorded.
(ex. To play recorded calls, check the box next to a 'Listen' icon and click 'Listen'.
Browser will prompt you to open the sound file in your favorite audio player or to
download the sound file).
(Option button)
Print
Print
Check the box next to a call record and click the 'Print' button. This action will open a
new popup window with the printing interface.
Email
Click on the 'Email' button to send all reports listed on the page or select a box next to
the report and click on the 'Email' button to send only the selected ones
Email
Provide an E-mail address where the report is to be sent and click on the 'OK' button to
proceed or 'Cancel' to abort the email action
Email
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Press 'OK' to email all CDR records on the current page (even if they are not selected)
or click 'Cancel' to print selected records only
Email
Finally, press the 'OK' button to confirm the email action or 'Cancel' to abort the email
action
Advanced
Advanced
CLIR
CLIR
• CLIR:
CLIR (Command Line Interface Record) details
192
(ex. Select a desired call record and click this button to view more technical details
about the call. A small popup window will open with the data.
NOTE: When experiencing any kind of unexplained problems, this is the data you need
to send to the technical support team)
(Command Button)
E-mail CLIR page option enables you to send a current CLIR to the desired e-mail
address
Delete Recording
• Delete Recording:
Deletes the recorded calls.

(ex. Select a recorded call and click this button to delete it from the file system)
(Command Button)
NOTE: For this command to be displayed, appropriate enhanced service has to be set.
Download CSV
• Download CSV:
Download data as the .csv (Comma Separated Value) file

(ex. Click this button to download the .csv file to your desktop)
(Command Button)
CDR summary
CDR summary is used when one wants to check the total cost of the calls that some
extension on the tenant made.
CDR summary
• Extension:
Extension for which system is presenting total cost of all calls.

(ex. 10001)
193
(Display)
• Cost:
Total cost of extension calls.

(ex. 10.66)
(Display)
Search/Filter
Search/Filter
• Start Date:
Select a Search/Filter start date

(Option button)
• End Date:

(Option button)
• Tenant(s):
Tenant(s) on which the calls happened.

(ex. 300)
([0-9])
• Acountcode(s):
Extension(s) which made calls.

(ex. 10001)
(Display)
TIP: After making any changes to the search filter, be sure to click the search icon
194
Actions
Print
new pop-up window with the printing interface.
Actions-Print
Email
Click on the 'Email' button to send all cost reports listed on the page or select a box next
to a report and click on the 'Email' button to send only the selected ones
Email
Provide the E-mail address where the report is to be sent and click on the 'OK' button to
Email
195
Press 'OK' to email all reports on the current page (even if they are not selected) or click
'Cancel' to print selected records only
Email
Finally, press the 'OK' button to confirm email action or 'Cancel' to abort the email
action
Download
• Download CSV:
Download data as a .csv (Comma Separated Value) file

(Command Button)
CDR settings
Please provide a number in 'Records per page' field
CDR settings
• Records per page:
196
Number of records displayed per page
(ex. When on the 'Reports: CDR' page, if this option is set to '16', the last 16 call
records will be displayed. On the bottom there is a 'Page' field. Type a page number,
e.g. '2', and click the 'GO' button to display the next 16 call records)
([0-9])
CLI Messages
CLI messages provide a convenient method of showing messages received from the
asterisk CLI (Command Line Interface). Each message is shown in the order received
and if clicked on, will open a new browser searching www.google.com with the
message content text.
CLI Messages
Available Message types:
• Warning - A warning message of an issue that will not usually affect the
system's operation
• Notice - A notice message is simply a formal notice and does not affect the
system's operation
• Error - An error message may in some situations stop or affect the system's
operation
SMTP Log
Last messages archived in the SMTP log. Messages are marked as:
• Sent - Sent by PBXware MT

• Received - Response from the SMTP server
197
SMTP Log
• Date:
Date/Time SMTP log was created

(12 Sep 2006 12:57:21)
(Display)
• Message:
SMTP server response

(AUTH LOGIN)
(Display)
Reports on slave tenants

Slave tenants can view CDRs of the calls that were made on them and can't view the
CDRs from the other tenants.
CDR
CDR displays detailed records of all calls on the current tenant with the following
details
CDR
• From:
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Extension number from which the call was made
(ex. If the call was made from extension 1001 to extension 1004, '1001' is displayed
here).
(Display)
• To:
Extension number to which the call was made

(Display)
• Date/Time:

(ex. 04 Oct 2006 10:44:10)
(Display)
• Duration:

(ex. 00:12:45)
(Display)
• Billing:

(ex. 00:12:45)
(Display)
• Status:

Example:
following content:
• Answered
• Not Answered
• Busy
• Error
(Display)
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service is active
Search/Filter
Search/Filter
• Start Date:

(ex. Click on the small 'Calendar' icon next to a field and select desired date)
(Option button)
• End Date:

(Option button)
• From:
Select whether you want to search CDRs by Destination(s) or Trunk from where the
call got in
(ex. Destination(s) or Trunks)
(Select box)
• To:
This field points to a Destination(s) or Trunk for which you are searching
(ex. Trunk)
(Select box)
• ID:
ID of the CDR. When a user selects a CDR, the ID field shows that CDRs ID. If there are
problems on the system, the customer can supply the support team with the
problematic CDR ID which helps in locating it.
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(ex. 1221447123.66)
([0-9] .)
• Start Time
When searching for CDRs this is the start time on the Start Date
([0-9] :)
• End Time
When searching for CDRs this is the end time on the End Date
([0-9] :)
• From
If you chose Destination(s) in the From Select box, you will enter the extension from
which the call came. If your selection was Trunk, you will have a Select box in this place
where you can choose a trunk on the system from which the call came.
Example:
• Destination(s) - 1009
• Trunk - Sales
([0-9] or Select box)
• To
Here you will enter number of the destination or select a trunk in which the call ended
up.
(ex. 1007)
([0-9])
• Status:

Example:
• All
• Answered
• Not Answered
• Busy
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• Error
(Select box)
TIP: After making any changes to search filter, be sure to click the search icon
Actions
Listen
• Listen:
Once the 'Listen' icon is displayed next to a call record it means that the specific call
was recorded.
(ex. To play recorded calls, check the box next to a 'Listen' icon and click 'Listen'.
Browser will prompt you to open the sound file in your favorite audio player or to
download the sound file).
(Option button)
Call
To establish a call between two extensions, all you need to provide is the caller
$EXTENSION number and the $DESTINATION extension
Call
• Caller
Extension that will make a call

Example:
Provide any extension number here, 1001, for example
([0-9])
• Destination:
Destination extension that will be dialed by the 'Caller' extension

(ex. To select a destination extension, first check the box next to a CDR record. This
field will display two extensions listed under 'From' and 'Destination' selected record)
(Select button)
TIP: After setting 'Caller' and 'Destination' extensions, click the call icon
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Print
new pop-up window with the printing interface.
Print
Email
Click on the 'Email' button to send all reports listed on a page or select a box next to a
report and click the 'Email' button to send only selected ones
Email
Provide an E-mail address where the report is to be sent and click on the 'OK' button to
Email
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Press 'OK' to email all CDR records on the current page (even if they are not selected)
or click 'Cancel' to print selected records only
Email
Finally, press the 'OK' button to confirm an email action or 'Cancel' to abort the email
action
Advanced
Advanced
CLIR
CLIR
• CLIR:
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CLIR (Command Line Interface Record) details
Example:
Select a desired call record and click this button to view more technical details about
the call. A small popup window will open with the data.
NOTE: When experiencing any kind of unexplained problems, this is the data you need
to send to the technical support team
(Command Button)
E-mail CLIR page option enables you to send the current CLIR to desired e-mail
address
Delete Recording
• Delete Recording:
Deletes the recorded calls. NOTE: For this command to be displayed, appropriate
enhanced service has to be set.
(ex. Select a recorded call and click this button to delete it from the file system)
(Command Button)
Download CSV
• Download CSV:
Download data as a .csv (Comma Separated Value) file

(Command Button)
CDR settings
CDR settings
Please provide a number in the 'Records per page' field
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• Records per page:
Number of records displayed per page

(ex. When on the 'Reports: CDR' page, and this option is set to '16', the last 16 call
records will be displayed. On the bottom there is a 'Page' field. Type a page number,
e.g. '2', and click the 'GO' button to display the next 16 call records)
([0-9])
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MT 3.8.5 Statistics
Statistics can be used to generate detailed information on the number of calls made
during the day or an hour. It can also be shown on an extension basis, meaning that
when you enter all the desired search information, you will get a list with the number of
calls per extension on the system.
Contents
• 1 Daily
o 1.1 Daily Statistics
o 1.2 Complete List of Calls
• 2 Hourly
o 2.2 Number of Calls per Hour
o 2.3 Month Range Charts
• 3 Extensions
o 3.2 Calls pre Extension Chart
Daily
Daily statistics will list the number of calls on a daily basis. This will show a list of days
with the number of calls which were made during those days.
Daily
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Clicking on the "Advanced Search" button will expand the above search and give users
more criteria to choose from:
Daily
• Date range, From:
This is the start date of the range of displayed calls

(ex. Mar-30-2010)
(Date picker)
• Date range, To:
This is the end date of the range of displayed calls

(ex. Apr-15-2010)
(Date picker)
• From
Number(s) from which calls shown are made. To show calls made from more than one
number, list all desired numbers as comma separated. By default, the "ALL" modifier is
in this field.
(ex. To display calls made from extension 104 and remote number 44207296666 you
would type "104,44207296666" without " ".
([0-9],)
NOTE: Depending on the select box right next to this field, you will be able to show
calls containing, beginning with, ending with or having the exact same number(s) as
• To
Number(s) to which calls shown are made. To show calls made to more than one
number, list all desired numbers separated by commas. By default, the "ALL" modifier
is in this field.
(ex. To display calls made to extension 104 and remote number 44207295555, you
([0-9] ,)
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calls containing, beginning with, ending with, or having the exact same number(s) as
• CallerID
CallerID from which the calls are shown.

(ex. "Tech Support" <064567876>).
([0-9][a-z] " < >)
calls containing, beginning with, ending with, or having the exact same CallerID as
provided in this field
• Trunk
Show the calls filtered on trunk basis.

(ex. First select box is used to select technology of the outgoing trunk. Second field is
used to enter trunk name for which outgoing calls should be shown. On second select
box, select whether the string contains, begins with, ends with or contains exact same
match).
([0-9][a-z])
• Duration (sec)
In these fields you can set a range for the duration of the calls in seconds. Also, both
fields can be modified depending on the equality signs in select boxes like: >, >=, ==,
<=, <
(ex. 300 with > would mean to show all calls with duration greater than 5 mins (300
seconds))
([0-9])
TIP:
Depending on the selection in the search section, two or three new sections with data
will be opened under the search.
If the date range selected is only one day, the third section will display the same graph
that you get when you click on the daily statistics breakdown which will show the
number of calls per hours of that day.
Download CSV option is used to download CSV formated file with all the statistics
shown on this page.
Daily Statistics
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This table will show the total number of calls per days in the date range selected in the
search box.
Daily Statistic
• DATE
Clickable date for which the total number of calls is shown
• TOTAL TIME
Total time of calls on that particular day
• TOTAL CALLS
Total number of calls made on that particular day
• CALL TIME
Average call time of the calls on that particular day
• GRAPHIC
This field shows the graphic representation of all calls on that day, relative to the day
that had biggest total call time. Day which had biggest total call time will have 100%
bar filling this field and every other day will have a bar relative to this day.
TIP:
TOTAL line on the bottom of the table will show the total times of all calls in this date
range, total number of calls of all calls, and the average call time of all calls in this date
range.
Daily Statistics Breakdown
Clicking on one day of daily statistics will bring up a new window with an hourly
breakdown of calls for that day
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• TIME
Hour for which the call statistics are shown
• TOTAL TIME
Total time of calls in that particular hour
• TOTAL CALLS
Total number of calls made in that particular hour
• AVERAGE CALL TIME
Average call time of the calls in that particular hour
TIP: The Download CSV option is used to download the CSV formatted file with the
hourly breakdown data.
The above picture shows a more graphic representation of the calls per hours on the
given day, and below is the list of all calls that were made on that day.
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• From
Number from which the call was made
• To
Number to which the call was made
• Date/Time
Date and Time on which the call was made
• Duration
Duration of the given call
Complete List of Calls
Below are the Daily Statistics: a complete list of all the calls in the searched date range
is shown.
Complete List of Calls
• From
• To
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• Date/Time
• Duration
Hourly
Hourly
Hourly statistics will list the number of calls on a daily basis but will provide more
detailed data on an hourly basis.
Clicking on the "Advanced Search" button, will expand the above search and give users
Hourly

(ex. Mar-30-2010)
(Date picker)
• Date range, To:

(ex. Apr-15-2010)
(Date picker)
• Month range, From:
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If you want to set a range or compare months of the current year, this is the starting
month of the range or comparison
(ex. Feb)
(Select box)
• Month range, To:
This is the ending month of the range or comparison. Date range and Month range
options are mutually exclusive which means that you can either use one or the other.
(ex. Apr)
((Select box)
• Compare
Depending on the selection of this option, the results will show data for the selected
range or will compare two days or two months of call statistics.
(ex. Selected, Range)
(Option buttons)
• To
Number(s) to which the calls shown are made. To show calls made to more than one
number, list all desired numbers separated by commas. By default, "ALL" modifier is in
this field).
(To display calls made to extension 104 and remote number 44207295555, you would
type "104,44207295555" without " ".
([0-9] ,)
calls containing, beginning with, ending with or having the exact same number(s) as
provided in this field)
• From
Number(s) from which calls shown, are made. To show calls made from more than one
number, list all desired numbers as comma separated. By default, the "ALL" modifier is
in this field.
([0-9] ,)
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provided in this field.
• CallerID

(ex. "Tech Support" <064567876>
([0-9][a-z] " < >)
• Trunk
Show the calls filtered on a trunk basis.

(ex. The first select box is used to select technology of the outgoing trunk. The second
field is used to enter a trunk name for which outgoing calls should be shown. On the
second select box, select whether the string contains, begins with, ends with, or
contains the exact same match).
([0-9][a-z])
• Duration (sec)
In these fields, you can set a range for the duration of the calls in seconds. Also, both
fields can be modified depending on the equality signs in select boxes like: >, >=, ==,
<=, <
(ex. 300 with > would mean to show all calls with duration greater than 5 minutes (300
seconds))
([0-9])
NOTE:
Depending on the selection in the search section, different tables and graphs with data
will be presented to the user.
The Download CSV option is used to download the CSV formatted file with all the
statistics shown on this page.
Daily Statistics
215
If the date range is selected in the search box and the Compare option is set to Range,
the following table with data on calls per day in that range will be shown. NOTE: If the
Compare option is set to Selected, the following table will show only two entries for the
days selected in the Date range.
Daily Statistics
• DATE
Clickable date for which the total number of calls isshown
• TOTAL TIME
Total call time on that particular day
• TOTAL CALLS
• CALL TIME
• GRAPHIC
This field shows a graphic representation of total calls on that day, relative to the day
TIP: TOTAL line on the bottom of the table will show the total times of all calls in this
date range, total number of calls of all calls and average call time of all calls in this date
range.
Clicking on one day on daily statistics will bring up a new window of hourly breakdown
of calls for that day
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• TIME
• TOTAL TIME
Total call time on that particular hour
• TOTAL CALLS
Total number of calls made on that particular hour
Average call time of the calls on that particular hour
TIP: The Download CSV option is used to download CSV formatted file with the
hourly breakdown data.
The picture shows a more graphic representation of the calls per hours on the given day,
and below is the list of all calls that were made on that day.
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• From
• To
• Date/Time
• Duration
Number of Calls per Hour
When the Date range is selected as an option, this chart will show the number of calls
on an hourly basis where one line represents one day from the range. If the Compare
option is set to Selected, this chart will show only two lines representing two days from
the Date range fields, and show calls during those days on hourly basis.
Number of Calls per Hour
TIP: One important fact about this chart is that whatever the range in Date range fields
is set, the maximum number of days presented on the chart will be ten.
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Month Range Charts
When the Month range option is selected in the search box, the user will be presented
with two pie charts presenting the total number of calls and total call duration per
month.
Month Range Charts
Extensions
Extension statistics will list the number of calls per extension. This will show a list of
days with the number of calls which were made during those.
Extensions
Clicking on the "Advanced Search" button will expand the above search and give users
Extensions
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(Mar-30-2010)
(Date picker)
• Date range, To:

(ex. Apr-15-2010)
(Date packer)
• Compare
Select whether to show calls that were dialed from extension(s) or calls that were
received to extension(s)
(ex. Dialed)
(Option buttons)
• To
Number(s) to which the shown calls, are made. To show calls made to more than one
number, list all desired numbers separated by a comma. By default, the "ALL" modifier
is in this field.
(ex. To display calls made to extension 104 and remote number 44207295555 you
([0-9] ,)
• From
Number(s) from which the calls shown are made. To show calls made from more than
one number, list all desired numbers as comma separated. By default, the "ALL"
modifier is in this field.
([0-9] ,)
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• CallerID

(ex. "Tech Support" <064567876>
([0-9][a-z] " < >)
• Trunk
Show the calls filtered on trunk basis.

(ex. The first select box is used to select the technology of the outgoing trunk. The
second field is used to enter trunk name for which outgoing calls should be shown. On
second select box, select whether the string contains, begins with, ends with or
contains exact same match).
([0-9][a-z])
• Duration (sec)
In these fields you can set a range for duration of the calls in seconds. Also, both fields
can be modified depending on the equal signs in select boxes like: >, >=, ==, <=, <
(ex. 300 with > would mean that all calls with duration greater than 5 minutes (300
seconds))
([0-9])
TIP:
The Download CSV option is used to download the CSV formatted file with all the
statistics shown on this page.
Daily Statistics
This table will show the total number of calls per days for every extension matching
criteria in the search box.
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Daily Statistics
• Answered
Total number of answered calls for this extension
• Unanswered
Total number of unanswered calls for this extension
• Duration < min
Total number of calls for this extension with call duration less than a minute
• DATE
The clickable date for which the total number of calls is shown
• TOTAL TIME
Total time of calls on that particular day
• TOTAL CALLS
• CALL TIME
• MAX CALL TIME
Maximum call time during this day
• Graphic
This field shows a graphic representation of all calls on that day, relative to the day
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TIP:
The TOTAL line on the bottom of the table will show the total time of all calls in this
date range, total number of calls, and the average call time of all calls in this date
range.
Clicking on one day of daily statistics will bring up a new window with an hourly
breakdown of calls for that day
• TIME
• TOTAL TIME
Total time of calls on that particular hour
• TOTAL CALLS
Total number of calls made on that particular hour
Average call time of the calls on that particular hour
TIP:
The Download CSV option is used to download the CSV formatted file with the hourly
breakdown data.
223
The above picture shows a more graphic representation of the calls per hour on the
given day, and below is the list of all calls that were made on that day.
• From
• To
• Date/Time
• Duration
Calls pre Extension Chart
This chart will show the number of calls made by or received by extensions in the date
range selected in the search box.
224
Calls pre Extension Chart
TIP: This chart, however, will show a maximum of ten extensions with the number of
calls in a given date range.
225
MT 3.8.5 Fax
The fax window displays all faxes received by PBXware and the ones transferred to
remote systems as well
Fax
Fax
This screen lists all faxes received by PBXware with the following details
Fax List
• Delete
With the Delete button you will remove the selected fax from the list
(Button)
• Download PDF
Download the selected fax as a PDF file

(Button)
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• Download TIFF
Download the selected fax as a TIFF file

(Button)
• From:
Extension number from which the fax was sent

(ex. 032445231)
• Destination:
The Email address to which the attached fax was sent. If the extension number is
displayed here, the fax is sent to the email address associated with the extension
(ex. [email protected]|1001)
(Display)
• Date/Time:
Date/Time fax was received

(ex. 04 May 2007 12:48:12)
(Display)
• Pages:
Number of pages in the received fax

(ex. 1)
(Display)
• Size:
Fax size in KB
(ex. 14KB)
(Display)
• Sent:
Shows whether the fax was sent remotely or not

(ex. The red icon indicates local and the green one indicates the remote fax
destination)
(Display)
• Box used for download/delete fax actions
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(ex. Select this box and click 'Download' button to download the selected fax)
(Option button)
Remote FAX
These options allow PBXware to transfer all incoming faxes to other systems. In order
to do so, be sure to set the following options under the incoming DID
('Destination'='Fax to Email', 'Value'='remote:fax') and then provide the necessary
remote system information here as follows
Remote Fax
• Remote PBXware:
The IP address of the remote system that is to receive the fax

(ex. 192.168.8.253)
([0-9])
• Remote Port:
Port on remote system used for communication

(ex. 10001)
([0-9])
• PBXware Username:
Remote system daemon username ('Settings: Servers: Edit: Daemon Username')

(ex. admin)
([0-9][a-z])
• PBXware Password:
Remote system daemon password ('Settings: Servers: Edit: Daemon Password')

(ex. pasd7f9)
([0-9][a-z])
• Remote Fax Number:

228
Extension number that will be displayed as fax sender.
(ex. 032445231)
([0-9])
229
MT 3.8.5 System
Contents
• 1 System
o 1.1 On master tenant
o 1.2 On slave tenant
System
System window administers the core PBXware MT components such as the file system,
system services, server details, licensing, sound files, MOH (Music On Hold) etc.
System
On master tenant
File System
All PBXware MT logs, sound recordings, CLI and CLIR files are stored on local file
system. Some of these files can grow to a size which will not leave any space left on the
system. This section provides management of how and when these files should be
rotated or deleted in order to prevent such scenario.
230
File System
Following files are rotated after selected period of time
• PBXware MT logs:
Time PBXware MT logs are kept on the filesystem

'/opt/pbxware/pw/pbxware/var/log'
(ex. Default value '4 weeks')
(Select box)
• Asterisk logs:
Time Asterisk logs are kept on the filesystem

'/opt/pbxare/pw/var/log/asterisk'
(Select box)
Following files are deleted after selected period of time
• PBXware MT CLIR files:
Time PBXware MT CLIR files are kept on the filesystem

'/opt/pbxware/pw/pbxware/var/clir'
(Select box)
• Asterisk Backup fiels:
Time Asterisk backup files are kept on the filesystem

'/opt/pbxare/pw/etc/asterisk/backup'
(Select box)
• Voicemail:
Time Voicemail files are kept on the filesystem

'/opt/pbxare/pw/var/spool/asterisk/voicemail/default/$VOICE MAILBOX'
(Select box)
• Recordings:
Time Recordings are kept on the filesystem
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'/opt/pbxare/pw/var/spool/ast/monitor'
(ex. Default value 'week')
(Select box)
Services
Services
This window controls the basic actions (start, stop, restart, reload) of PBXware MT
services
• System:
Stop/Start/Restart the system

(ex. Clicking on the 'Restart' button would reboot the system)
(Command button)
• PBX service:
Stop/Start/Restart/Reload the Asterisk (The core PBXware MT runs on)

(ex. Clicking on the 'Restart' button would restart the Asterisk)
(Command button)
• PBXware MT:
Stop/Start/Restart/Reload the PBXware MT

(ex. Clicking on the 'Restart' button would restart the PBXware MT)
(Command button)
• HTTP service:
Stop/Start/Restart/ web server

(ex. Clicking on the 'Restart' button would restart the web server. In case you cannot
reach the PBXware MT login screen this is the service you need to start/restart)
(Command button)
232
• Database service:
Stop/Start/Restart/Reload database service

(ex. Clicking on the 'Restart' button would restart the database server)
(Command button)
• TFTP service:
Stop/Start/Restart/ the TFTP server

(ex. TFTP is used for storing and serving the UAD/Phone auto-configuration files)
(Command button)
• Jabber server:
Stop/Start/Restart/Reload the Jabber messaging server

(ex. Clicking on the 'Restart' button would restart the Jabber server)
(Command button)
Server Details
Server Details
This window resets the root PBXware MT password, timezone, and hostname
• Root Password:
PBXware MT prompts for this password during the system/ssh login and when
accessing system services through the interface
([a-z][0-9])
• Confirm Password:
Re-type the Root Password entered in the above field

([a-z][0-9])
• Time zone:
233
Select the appropriate time zone ni which, PBXware MT is located
(Select box)
• Hostname:
(ex. myhost)
([a-z][0-9])
For information on the rest of the fields in the Server detail, check Getting Started-
>Server Details chapter
Licensing
This window (re)licenses the system. Free and requested licenses are available (both
bond to a system MAC address). If you are upgrading your license just paste the license
key in 'License Number' field, select the MAC address assigned to a license key and
click on the 'Save' button
TIP: System must have access to a fully operational Internet connection in order to
license the system.
Licensing
• License Type:
Select a system license type

(ex. The system comes with two license types. FREE license and 'Requested License')
(Select box)
• License Number:
Provide system license number as received in email.

(ex. ABCDE123)
([a-z][0-9])
NOTE: This field will not be active if FREE license is requested

234
• MAC:
Select a MAC address to which the license will be applied

(ex. If your system has multiple MAC addresses, select the one you wish to assign the
license to. In case this MAC address changes in the future, you will have to re-license
the system)
(Select options)
Sound Files
Sound Fields
This section administers global sound files used by all of the slaves on PBXware MT,
which means that all extensions from all of slave are using these as their general sound
files, i.e. 'Please enter the conference PIN number'
• Sounds:
Select among available options (gsm, ulaw, alaw, g729, ilbc, sln) to display system
sound files of that type
(ex. Selecting 'gsm' will display all system sounds with the .gsm file type)
(Select box)
• Upload:
Uploads the selected file from the local computer to PBXware MT

(ex. Click on the 'Browse' button and select a file from your desktop. Then click this
button to upload the selected file to PBXware MT)
(Command button)
• Convert:
235
Convert the selected file to the desired codec.
(ex. Select the box next to a sound file (e.g. arizona). Select a codec from the drop
down menu and click on Convert. The sound file will be converted to that codec).
(Command button)
• Rename:
Renames the selected sound file

(ex. Select the box next to a sound file (e.g. arizona). Change 'arizona' into 'Arizona101'
and click this button to rename the selected sound file)
(Command button)
• Delete:
Deletes the selected sound file

(ex. Select the box next to a sound file (e.g. arizona) and click this button to delete the
selected sound file)
(Command button)
• Download:
Downloads the selected sound file to the user's desktop

(ex. Select the box next to a sound file (e.g. arizona) and click this button to download
the selected file to your desktop)
(Command button)
TIP: PBXware MT will play only sound file types equal to enabled codecs on dialing
extensions.
For example, Extension 1000 has only gsm codec enabled. When the same Extension
logs in as a Queue Agent by dialing '*202 + $AGENT_NUMBER', all sounds played by
PBXware MT (asking for password etc.) will be in '.gsm' format.
If multiple codecs are enabled for Extension 1000(ulaw,alaw,gsm), PBXware MT will

play sound files with better sound quality (ulaw/alaw).
Music on Hold
Music on Hold
236
Music on Hold is music or advertisements played to callers while they are waiting for
an agent or when put on hold, for example. MOH set on master tenant is played if there
is none matched on a slave where MOH is supposed to be played
Content
MOH content window administers PBXware MT default MOH sound files. They are
listed here with the following details
• Name:
MOH sound file name

(ex. Song)
(Display)
• Author:
MOH sound file author

(ex. Jinkies)
(Display)
• Length:
MOH sound file length in min:sec

(ex. 3:08)
(Display)
• Class:
MOH class sound file belongs to

(ex. Default)
(Display)
• Status:
MOH sound file status

(ex. On/Off)
(Display)
• Edit
Edits the MOH file configuration

(ex. Click to edit MOH file configuration)
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(Button)
• Delete
Deletes a MOH file from the system

(ex. Click to delete a MOH file from the system)
(Button)
Search
Search
• Search:
Search phrase
([a-z][0-9])
• Name:
Should a search filter be applied to track names

(ex. Check the box to search track names)
(Check box)
Add Record
Add Record
• Clip name:
Audio file name

(ex. Riders on the Storm)
([a-z][0-9])
• Author:
238
Author name
(ex. The Doors)
([a-z][0-9])
• File:
Displays full path to music file on local computer. Click the 'Browse' button to select a
file
(ex. D:\Music\The Doors\The Doors - Riders on the Storm.mp3)
([a-z][0-9])
• Length:
Clip length
(ex. 4:13)
([0-9][:])
• Class:
Select the MOH class to which the sound file belongs

(ex. default)
(Select box)
• Status:
Set the status of uploaded file(active/inactive)

(ex. On/Off)
(Option buttons)
Sangoma cards
This part of the interface is used to set various Sangoma cards: PRI, Analog and BRI.
Sangoma cards
239
Depending on the cards inserted and the number of ports on them, appropriate row of
information will show up in the list of the cards. Also whether a PRI, Analog or BRI
card is inserted, they will have different configuration settings.
PRI cards
When a PRI card is inserted (like A101 which has one port) you will get as many
"items" in the list as you have ports on that particular card. In this case, A101 has one
port hence one item or card in the list. If you insert A102 you will get two items in the
list because that card has two ports.
So, you need to configure every port individually, that's why there is a separate
Configure button by every port.
PRI cards
When the card is inserted you will see 'Not connected' Status. You need to connect the
cable to it and click on Configure button.
PRI cards
• Media type:
This is the type of the PRI system that the card is connecting to
Example:
• T1 - PRI in US
• E1 - PRI in EU
• J1 - PRI in Japan
(Select box)
• Coding:
240
This is the coding type for this port. This is automatically selected with Media type, so
there is no need to change this option.
(ex. B8ZS)
(Select box)
• Framing:
This is the framing used for this port. This is automatically selected with Media type, so
there is no need to change this option.
(ex. ESF)
(Select box)
• Clock:
Select whether the card will give a clock to devices (Master), or use a clock generated
from telco (Normal)
(ex. Master)
(Select box)
• Reference clock:
This option allows one to use the incoming clock from a different port as a clock source
to this port.
(ex. 2)
([0-9])
NOTE: This option works only if Clock is Master.
• Hardware DTMF:
Enable/Disable Hardware DTMF detection

(ex. Yes, No, N/A)
(Option buttons)
When you choose which type of connection is used and whether the card will use a
clock source from the provider or generating one, click on the Save button. After that
click on Apply and then the Restart PBXware MT button. If the cable is connected and
configuration done properly, the system will display 'Connected' by the cards port.
Analog Cards
241
Analog Cards
When you insert an analog Sangoma card, each card has one "item" in the list,
regardless of number of ports that it has on it. Under the Port, it will show you how
many FXS and/or FXO ports are configured.
Analog Cards
• Codec:
Codec used for this card

(ex. MULAW)
(Select box)
• FXO Operation mode:
FXO Operation mode.

(ex. FCC)
(Select box)
• Hardware DTMF:
Enable/Disable Hardware DTMF detection

(ex. Yes, No, N/A)
(Option buttons)
The procedure for configuring FXS/FXO ports is simplest. Click on Configure, Save,
Apply and then Restart PBXware MT.
TIP: If you change the hardware configuration of the card (PCI slot, number of
FXS/FXO modules etc.) first reset the settings by clicking on the Reset button in the
Configuration window, and then Save.
242
BRI Cards
BRI Cards
Here you can configure Sangoma BRI cards. In this example we used A500 which has 3
physical ports where every port splits into two ISDN ports by a special cable, thus
offering 6 ISDN ports which you must configure by clicking on the Configure button.
BRI Cards
• Connection type:
This should match the settings provided by Telco.

(ex. point-to-point)
(Select box)
• Group:
Select in which group you want to place the current port. This is useful if you want one
trunk to use more than one port so all calls could be spread across several ports. In
such a case, put the same number on all ports you want to allocate to the trunk.
(ex. 2)
([0-9])
• Country:
Set this to your local country.

(ex. usa)
(Select box)
• Number of incoming digits:
243
The minimum number of digits for a called number on an incoming call. Incoming calls
with a called number less than this value will be rejected with cause: Invalid number
format.
(ex. 6)
([0-9])
• Dial plan:
Some switches require the "type of number" network specific facility to be set.
(ex. international)
(Select box)
• Numbering plan:
Some switches require the "number plan" network specific facility to be set.
(ex. isdn)
(Select box)
If the port was connected and properly configured, after clicking on Save and then
Apply, restarting PBXware MT will show Connected by that port.
Information
PBXware MT continuously monitors system services, load, ZAPTEL modules etc... in

order to achieve and maintain high quality operations across the system and network.
The result of these monitoring operations are also accessible to an authorized user.
Information
• Refresh Interval:
Time interval in seconds at which data is to be refreshed

(ex. 10 sec)
244
(Select box)
• PBXware MT Uptime:
Total time PBXware MT is available for service

(ex. 1 hour, 16 minutes, 42 seconds)
(Display)
• Last Reload:
Total time since PBXware MT was last reloaded

(ex. 10 minutes, 34 seconds)
(Display)
• Load:
Load shown for past 1, 5 and 15 minutes

(ex. 0.16)
(Display)
• CPU:
CPU usage by: user, kernel and idle

(ex. 2.3%)
(Display)
• Memory:
Memory usage by: Used, Cache and Free

(ex. 299.1M)
(Display)
• Swap:
Swap space usage

(ex. 5.2M)
(Display)
• Processes:
Processes by Running, Sleeping, Stopped and Zombie

(ex. 1, 96, 0, 0)
245
(Display)
• rootfs:
File systems present shows by type, mount, usage and free status
(ex. 0B, 0B, 3.3G, 1.5G)
(Display)
• CPU Info:
Number of CPU's, Model, Speed and Cache size

(ex. Core 2 Duo, 1.6Ghz, 2MB)
(Display)
• System:
General system details like Name, Kernel, Architecture and Uptime

(ex. zenica.domain.com, Linux 2.6.31-27.0.1.EL, i686, 18d 1h 35m)
(Display)
• Services:
Default system services running on the system

(ex. PBXware MT, Asterisk, TFTP, Postfix)
(Display)
• Modules:
Currently loaded ZAPTEL modules

(ex. wsusb)
(Display)
• Zaptel:
A list of all cards detected by the system is displayed here.

Channel Map displays used slots on TDM card. In this case the first slot is filled with
FXO module (displayed in black) while other 'Empty' slots are displayed in gray color.
(ex. Channel Map: 1: FXO, 2:FXS, 3:Empty, 4:FXO)
(Display)
TIP: With TDM cards, please make sure your power cable is connected. The message
'PLEASE CHECK TDM POWER CABLE' will be displayed if this happens.
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On slave tenant
Sound Files
This section administers sound files used only by the current tenant. This is used to
manage only greeting sound files, and overriding of the system's default sound files is
not allowed.
Sound Files
• Sounds:
Select among available options (gsm, ulaw, alaw, g729, ilbc, sln) to display system
sound files of that type
(ex. Selecting 'gsm' will display all system sounds with the .gsm file type)
(Select box)
• Upload:
Uploads the selected file from a local computer to PBXware MT

(ex. Click 'Browse' button and select a file from your desktop. Then click this button to
upload selected file to PBXware MT)
(Command button)
• Convert:
Convert selected file to desired codec.

(ex. Select a box next to a sound file (e.g. arizona). Select a codec from the drop down
menu and click on Convert. The sound file will be converted to that codec).
(Command button)
• Rename:
Renames the selected sound file

(ex. Select a box next to a sound file (e.g. arizona). Change 'arizona' into 'Arizona101'
and click this button to rename the selected sound file)
247
(Command button)
• Delete:
Deletes the selected sound file

(ex. Select a box next to a sound file (e.g. arizona) and click this button to delete the
selected sound file)
(Command button)
• Download:
Downloads selected sound file to user's desktop

(ex. Select a box next to a sound file (e.g. arizona) and click this button to download
selected file to your desktop)
(Command button)
TIP: PBXware MT will play only sound file types equal to enabled codecs on dialing
extension.
For example, Extension 1000 has only gsm codec enabled. When same Extension logs
in as a Queue Agent by dialing '*202 + $AGENT_NUMBER', all sounds played by
PBXware MT (asking for password etc.) will be in '.gsm' format.
If multiple codecs are enabled for Extension 1000(ulaw,alaw,gsm), PBXware MT will

play the sound files with better sound quality (ulaw/alaw).
Music on Hold
Music on Hold is music or advertisements played to callers while they are waiting for
an agent or when put on hold, for example.
Music on Hold
Content
MOH content window administers PBXware MT default MOH sound files. They are
listed here with the following details
• Name:
MOH sound file name
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(ex. Song)
(Display)
• Author:
MOH sound file author

(ex. Jinkies)
(Display)
• Length:
MOH sound file length in min:sec

(ex. 3:08)
(Display)
• Class:
MOH class sound file belongs to

(ex. Default)
(Display)
• Status:
MOH sound file status

(ex. On/Off)
(Display)
• Edit
Edits the MOH file configuration

(ex. Click to edit MOH file configuration)
(Button)
• Delete
Deletes a MOH file from the system

(ex. Click to delete a MOH file from the system)
(Button)
Search
249
Search
• Search:
Search phrase
([a-z][0-9])
• Name:
Should the search filter be applied to track names

(ex. Check the box to search track names)
(Check box)
Add Recording
Add Recording
• Clip name:
Audio file name

(ex. Riders on the Storm)
([a-z][0-9])
• Author:
Author name:
(ex. The Doors)
([a-z][0-9])
• File:
Displays full path to music file on local computer. Click 'Browse' button to select a file
(ex. D:\Music\The Doors\The Doors - Riders on the Storm.mp3)
([a-z][0-9])
250
• Length:
Clip length
(ex. 4:13)
([0-9][:])
• Class:
Select a MOH class sound file belongs to

(ex. default)
(Select box)
• Status:
Set the status of uploaded file(active/inactive)

(ex. On/Off)
(Option buttons)
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MT 3.8.5 Routes
Contents
• 1 Routes
o 1.1 Routes
o 1.2 Destination Groups
o 1.3 Disabled Routes
Routes
Routes identify each number dialed by users. Information is identified by the number
dialed and the destination group it belongs to.
Routes
252
Routes
This window identifies all destinations by dialed number. Destinations that do not exist
in this database cannot be dialed by the PBXware MT.
Depending on what is selected when you edit the Routing mode on the master tenant in
Settings->Tenants menu, you will have two routing modes:
• E.164 Routing mode

• Simple Routing mode
E.164 Routing mode
When you are using E.164 Routing mode all destinations are defined by E.164 standard.
E.164 Routing mode
• Update daemon database:
Updates daemon database with imported data

(ex. Click to update)
(Button)
• Import Database:
Imports current data from the central destination database

(ex. Click to import)
(Button)
• 'Export database:
Export daemon database with newly added details

(ex. Click to export)
(Button)
253
TIP: Import and update operations need to be performed at regular maintenance times.
In addition, the network administrator should be notified of any destination that is not
accessible.
• Routes:
Displays all available destination routes alphabetically.

(ex. A, B, C, D...)
(Button)
• Destination group:
Further description of selected route. Displays destinations by proper, mobile etc

number
Example:
• Special Service
• 48 States
• Toll Free
• Alaska
• Hawaii
(Display)
• Show Hidden Groups:
Show hidden route groups

(ex. Click on the button to display all available groups)
(Button)
In order to view destination numbers for the United Kingdom, click on the 'u' letter
under 'Routes' navigation. All United Kingdom destination groups will be displayed.
Click on one (e.g. Proper) to view all destination numbers assigned to it.
Routes
Add/Edit Routes
You will see these options if you have clicked on the 'Add Destination' group or 'Edit'
icon under the routes list
254
Add/Edit Routes
• Destination Name:
Unique destination name

(ex. All proper destinations)
([a-z][0-9])
• Destination Code:
Destination code identifier

(ex. Number that identifies the destination (e.g. 442, where all numbers start with
442*******))
([0-9])
• Route:
Route to which the destination code belongs

(ex. If route '442' belongs to the United Kingdom, select 'United Kingdom' here)
(Select box)
• Destination Type:
Group to which destination code belongs

(ex. If route '442' belongs to proper phones, select 'Proper' here. If it belongs to mobile
provider O2 for example, select 'O2' here)
(Select box)
Simple Routing mode
This routing mode doesn't require numbers to be entered in the E.164 form, but this
mode allows for greater flexibility when used right.
255
Simple Routing mode
• Route
Name of the route

(ex. Mobile)
(Display)
• Start digits
First digits by which the system would recognize a call.

(ex. 06)
(Display)
• Required Length
Full length of dialed number recognized.

(ex. 9)
(Display)
• Regex
Regular expression which can be used to fine tune which starting numbers to fall
under that route.
(ex. 06[123])
(Display)
• Prefix
Prefix which should be added to dialed numbers.

(ex. 0044)
(Display)
• Strip Digits
Whether the prefix by which the number was recognized, will be stripped when
dialing.
(ex. Yes)
(Display)
256
Add routes
By clicking on Add route you will be able to add a new simple route to the system.
Add routes
• Route Name:
Name of the route being added

(ex. Mobile)
([a-z][0-9])
• Start Digits:
Starting digits of the numbers which will fall under this Route. If you leave this field
empty, any number will be taken into account, but in that case you must select how
many digits those numbers will have.
(ex. 06)
([0-9])
• Required Length:
Length of dialed numbers which would be recognized as this route. If this field is set to
none, any length of numbers will fall under this route, but Start Digits must be set, for
this to work.
(ex. 9)
(Select box)
• Required Regex:
This is an optional regular expression which can be used to fine tune numbers
recognized by this route
(ex. 06[1-5] - In this example only numbers starting with 061, 062, 063, 064 or 065 will
fall under this route. Also you could use other regular expressions here).
([0-9])
257
• Prefix:
Prefix which should be added to dialed numbers.

(ex. 00)
([0-9])
• Strip Start Digits:
Whether to strip start digits from the dialed numbers.

(ex. Checked)
(Check box)
Operation Times
Set the system open/closed times. Depending on the time when the call is received, the
call can be redirected to different PBXware MT destinations.
Operation Times

(ex. Yes, No)
(Option buttons)
• PIN
If you have a PIN set here, the system will ask you for it so you could pass through
Operation Time
(ex. 3425)
([0-9])
• Greeting:
258
(ex. greeting-***)
(Select box)
Open days: Sets the working hours during which the DID is to redirect calls as set in
the DID Add/Edit window. If any call is received during the hours not set here, 'Range
Destination' is checked, and if they do not apply, the call is redirected to 'Default
Destination'.
Closed dates: Sets the specific date when all calls are redirected to 'Default
Destination'. If 'Destination' field in the Closed dates is set, call will not go to 'Default
Destination' but to this number.
Destination Groups
Each country can be assigned with many destination groups/service providers. You may
add one by clicking on the Add Destination Group from this location. All groups that
have no numbers assigned will not be displayed when Routes information is required
Destination Group
• Destination Group:
Name of the destination group

(ex. 48 States)
(Display)
• Delete
Deletes a destination group from the system

(ex. Click to delete a destination group from the system)
(Button)
259
Add/Edit Destination Group
You will see this option if you have clicked on the 'Add Destination Group' option
Add/Edit Destination Group
Destination Group name

(ex. 48 States)
([a-z][0-9])
Disabled Routes
Disabled Routes
If you want some destinations to be disabled, you can add them to Disabled Routes,
separated by new lines if more than one is entered.
260
MT 3.8.5 LCR
LCR
The LCR (Least Cost Routing) section allows fine tuning the slaves trunks usage
accordingl to the price and quality. By default, the slave uses default trunks for each
destination in order to present this feature as simply as possible.
LCR
• Routes:
Displays all available destination routes alphabetically.

(ex. A, B, C, D...)
(Button)
Further description of selected route. Displays destinations by proper, mobile, etc.

number
Example:
• Special Service
• 48 States
• Toll Free
• Alaska
• Hawaii
(Display)
Select trunks to be used for the desired destination

(ex. Voip trunk)
(Select box)
Precedence
261
Settings:
• Default Trunks: All System calls go through the trunks defined here
• MiniLCR: Overrides 'Default Trunks' and sets a specific trunk for a destination
Extensions:
• Trunks: Overrides 'Settings: Default Trunks'

• Routes: Overrides 'Settings: MiniLCR'
262
MT 3.8.5 Service Plans
Service plan defines billing details for all available destinations but it will also enable
you to create a template for enhanced services, destinations and Online Self Care
settings that will be automatically applied to extensions associated with service plan.
Service Plans
Contents
• 1 Service Plans
o 1.1 Add/Edit Service Plan
• 2 Time Based Dialing
o 2.1 Add TBD rule
Service Plans
Service Plans
263
This screen lists all Service Plans with the following details
• Name:
Service plan name

(ex. Euro)
(Display)
• Edit
Edits the service plan

(ex. Click to edit a service plan configuration)
(Button)
• Delete
Deletes a service plan from the system

(ex. Click to delete a service plan from the system)
(Button)
Add/Edit Service Plan
These options fine tune the service plan with details such as minimum and connection
charge, grace period, and inclusive minutes
Add/Edit Service Plan
• Service Plan Name:
Service plan name

(ex. If service plan name is 'Euro', select this name under 'Service Plan' under
extensions to apply it to that Extension)
([a-z][0-9])
264
• TBD:
Should Time Based Dialing be applied to the current Service Plan

(ex. Yes, No, N/A)
(Option buttons)
• Minimum charge:
Minimum charge applied to each made call regardless of the call duration
(ex. If call is made, no matter how much it lasts, this 'minimum charge' will be applied)
([0-9])
• Connection charge:
Charge applied to any call that leaves the system (regardless if other party answers or
not)
(ex. If this charge is set to 0.4, each call that leaves the system will be charged that
amount regardless if other party answers the call or not)
([0-9])
• Total Inclusive Minutes:
Total number of inclusive minutes assigned to a service plan

(ex. If this field is set to '5', each user assigned to this service plan will have 5 free
minutes of call time to any location.
([0-9])
NOTE: 'Minimum charge' and 'Connection charge' will be applied if set.
• Grace Period:
Number of seconds at the beginning of a call that are not charged

(ex. If the grace period is set to 10, and a call lasted 15 seconds, only 5 seconds of the
call will be charged)
([0-9])
• Billing:
Billing type (MINIMUM_CHARGED/CHARGE_EVERY_$SECONDS)

(ex. If '30/6' is selected, and you've made the call which lasted 12 seconds, it will be
billed as if you've made a 30 seconds call. If the call lasted for 39 seconds - it will be
billed as if the call lasted 42 seconds (30 + 6 + 6 = 42))
(Select box)
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Rates
Rates
These options set incoming/outgoing rates per each known destination
Upload/Download
Destination Group rates can be easily uploaded and download from the server. Update
the .CSV file on your desktop, click on 'Browse' button, select the file and click on the
'Upload' button.
TIP: CSV file must be in following format

(Code,"Route","Destination","Outbound","Inbound"). For example
93,"Afghanistan","mobile"
To download rates file from the server just click on 'Download CSV' button.
Upload/Download
Routes
Click on a letter under a 'Routes' navigation 'A' for example and select 'Australia'. A list
of 'Destination Groups' will be displayed under 'Destination Group'. Click on the 'Edit'
button to edit the Destination Groups charges.
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Upload/Routes
TIP: If Time Based Dialing is turned on, you will enter charging for all TBD rules
Routes
• Outbound:
Destination group outbound charge

(ex. If you edit the 'Mobile' destination group and set this option to 5.00000, this rate
will be applied to all calls made to Mobile destination)
([0-9])
• Inbound:
Destination group inbound charge

(ex. If you edit the 'Mobile' destination group and set this option to 5.00000, this rate
will be applied to all calls received from this Mobile destination)
([0-9])
• Inclusive:
Should inclusive minutes be calculated for this destination

(ex. If this option is enabled, inclusive minutes will be applied when dialing or receiving
calls from this location)
(Option buttons)
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Enhanced Services
Enhanced Services
Enhanced Services set here will be applied to all users assigned to this Service Plan. For
example, if the 'Euro' Service Plan is set to have only 'Call Forwarding' enabled, all
users with the 'Euro' Service Plan will have 'Call Forwarding' enabled only.
For more on Enhanced Services, please click Enhanced Services
Destinations
Destinations
Destinations set here will be applied to all users assigned with this Service Plan. For
example, if the 'Euro' Service Plan is set to have the 'UK: Proper' destination allowed
only, all users with 'Euro' Service Plan will be able to call only 'UK: Proper'
Destinations
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For more on Destinations please click Destinations
Time Based Dialing

Time Based Dialing is a feature which enables call charging by specific TBD rules.
These rules specify the date/time for which they apply. When entering Rates in Service
Plans with enabled TBD, you will be able to enter price rates for every specified TBD
rule.
Time Based Dialing
• Rule name
TBD rule name

(ex. Working hours)
(Display)
• Priority
Priority of TBD rule

(ex. 55)
(Display)
• Edit
Edits the TBD rule

(ex. Click to edit the TBD rule)
(Button)
• Delete
Deletes the TBD rule

(ex. Click to delete the rule)
(Button)
Add TBD rule
These options are used for adding/editing specific TBD rules

269
Add TBD rule
• Rule name:
TBD rule name

(ex. Working hours)
([a-z][0-9])
• Rule priority:
This field defines priority of the TBD rule. If dates or times of two or more rules are
overlapping, the rule with the higher priority is taken.
(ex. 55)
([0-9])
NOTE: Priorities are numbers between 1 and 100, where 100 is the highest priority.
• Rule mode:
Depending on the rule mode, the TBD rule can be set by the Date range or the
Day/Time range
Example:
• Date/Time range
• Date range
(Select box)
• Day range:
Beginning day of the day range in which the TBD rule applies
(ex. Monday - Sunday)
(Select box)
• Day range:
Ending day of the day range in which the TBD rule applies
(ex. Monday - Sunday)
(Select box)
270
• Time range:
Beginning time of the time range in which the TBD rule applies
(ex. 00:00h - 23:00h)
(Select box)
• Time range:
Ending time of the time range in which the TBD rule applies
(ex. 00:00h - 23:00h)
(Select box)
• Date range:
Pick start and end date cliking on the buttons, in which the TBD rule applies
(ex. 09-03-2008)
(Buttons)
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MT 3.8.5 Settings
Contents
• 1 Settings
o 1.1 Settings: On master tenant
o 1.2 Tenants
o 1.3 Tenant packages
o 1.4 Protocols
o 1.5 Providers
o 1.6 E-mail Templates
o 1.7 Voicemail
o 1.8 Configuration Files
o 1.9 g729
o 1.10 About
o 1.11 Settings: On slave tenants
o 1.12 Default Trunks
o 1.13 UAD
o 1.14 Access Codes
o 1.15 Numbering Defaults
o 1.16 About
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Settings
Settings: On master tenant
Master Settings are used to set global MT variables. These settings are applied on the
entire system if not otherwise specified on a lower level.
For example: 'Settings: Voicemail: Send Attachment'='Yes' will set 'Set Attachment' for
all local and remote extensions. But each extension can override this rule by setting
'Extensions: Edit: Advanced Options: Attach'='No'
Settings: On master tenant
273
Tenants
PBXware MT system administration theoretically allows up to 800 slave tenants to be
administered from a single administration interface, but more than 100 is not
recommended.
Tenants
• Tenant Name:
Unique, custom set, tenant name.

(ex. Slave Tenant #2)
(Display)
• Package:
Name of the package used by a slave tenant.

(ex. Full package)
(Display)
• Edit
Edits the tenant configuration

(ex. Click to edit tenant configuration)
(Button)
• Delete
Deletes a tenant from the system

(ex. Click to delete a tenant from the system)
(Button)
274
Network Info
Network Info
• Server Name:
Set the master or slave tenant name here.

(ex. Slave #1)
([a-z][0-9])
• Jabber server ID:
ID on which Jabber users will log on.

(ex. voip.domain)
([a-z][0-9])
• Packages:
Tenant package that the slave is using:

(ex. Full package)
(Select box)
• Unique Tenant Code:
This is the unique code assigned to a slave tenant. When a phone is registering to an
extension, its username is set to TENANT_CODE+EXT_NUMBER. You can specify only
numbers between 200 and 999.
(ex. 200)
([0-9])
275
(ex. Click to edit network configuration)
(Button)
Server Details
Server Details
• Root Password:
Set the root password

(ex. PBXware MT prompts for this password during the system/ssh login and when
accessing system services through the interface)
([a-z][0-9])
• Confirm Password:
Confirm the root password

(ex. Re-type the Root Password entered in the field above)
([a-z][0-9])
• Time zone:
Time zone in which PBXware MT is located

(ex. Select the appropriate time zone, for example 'USA/East-coast)
(Select box)
• Hostname:
(ex. hostname)
([a-z][0-9])
General Settings
276
General Settings
• Default Server:
Which tenant is shown in the interface by default.

(ex. Select Yes to make a tenant default when showing the interface, which will set all
other tenants to No).
(Option buttons)
• Announce Trunks:
Announce over which trunk the call goes through

(ex. John dials 55510205 and this call goes over the secondary default system trunk.
John will hear 'Using the secondary trunk to terminate your call' message).
(Option buttons)
• Absolute Timeout:
Maximum time a call can last (in seconds)

(ex. If '3600' is set in this field, that will make all calls end after 1 hour ( 1h = 3600
seconds))
([0-9])
• Voicemail in CDRs:
Sets how the calls that were unanswered and redirected to voicemail are displayed in
CDR
(ex. A call was made to extension 1000 but was not answered. The caller gets
redirected to voicemail. If 'As Not Answered call' is set, CDR will display this voicemail
redirection as an 'Unanswered' call. If no option is set, the same call will be displayed
as 'Answered')
([0-9])
• Fax page mode:
Page format when downloading fax as a PDF
277
(ex. letter)
(Select box)
• Routing mode:
What type of routing mode will be used for Routes. (Only on master tenant)
Example:
• E.164 routing - routing mode where routes are set using E.164 numbering
rules
• Simple routing - routing mode where you manually set routes to fit your needs
(Select box)
• Default CallerID:
If set, calls that are going out through a trunk assigned to this tenant will have this
CallerID. (Only on slave tenants)
(ex. Tenant-001)
([0-9][a-z])
• Use Default CallerID for tenant to tenant calls:
Whether the Default CallerID will be used when making calls from the extension on
one tenant to extension on another tenant. (Only on slave tenants)
(ex. Select Yes to use Default CallerID).
(Option buttons)
• Default domain for OSC login:
Default domain used for OSC login.

(ex. http://pbx.domain.com/)
([a-z][0-9])
• Disable Tenant to Tenant calls:
When this option is enabled, extensions from one tenant will not be able to call
extension from another tenant which is done dialing TENANT+EXT
(ex. Yes, No, N/A)
(Option buttons)
• Disable CallerID rewrite for tenant to tenant calls:
278
If enabled, any call between tenants will not have overwritten CallerID. It will send
anything that extension has set on it.
(ex. Yes, No, N/A)
(Option buttons)
• Find E.164 numbers in DIDs:
Enable or disable E.164 matching of numbers as set in the E.164 field in DIDs
(ex. Yes, No, N/A)
(Option buttons)
• Number of objects per page:
Number of objects per page like extensions, DIDs and trunks.

(ex. 20)
([0-9])
• Show Directory in OSC:
Whether to show the Directory in Online Self Care. Only those extensions which allow
this, will be shown in Directory
(ex. Yes, No, N/A)
(Option buttons)
System operation time

(Option buttons)
Call Recordings
Call Recordings
• Record calls by default:
When creating an extension, Yes, No or N/A will be selected by default, depending on

the selection that you make here
(ex. Yes, No, N/A)
279
(Option buttons)
• Use MixMonitor:
MixMonitor allows you to record conversations with the possibility to adjust the heard
and spoken volume and to append the next conversation in the same file. So, at the
end of the day, you could have all the conversations on one channel in one file. They
will be stored in the same sequence, as they are made.
(ex. Yes, No, N/A)
(Option buttons)
• Silent recording by default:
When creating an extension, Yes, No, or N/A will be selected for Silent recording,
depending on the selection you make here
(ex. Yes, No, N/A)
(Option buttons)
• Email instant recording interval (min):
How often to send an email of instant recording in minutes. (Only on master tenant)
(ex. 3)
([0-9])
• Recordings format:
Format used for saving the system call and voicemail recordings. You can read more
details about disk space usage on the bottom of this chapter
(ex. Choose one of the following formats: gsm, wav, wav49 and ogg. If wav is selected,
all call recordings and voicemail recordings will be save in this format).
(Select box)
• Use RAM disk:
To speed up the process of recording you can turn on the RAM disk which practically
records all calls to RAM memory before storing them on the disk
(ex. Yes, No, N/A)
(Option buttons)
• RAM disk size:
Amount of memory used for call recording

(ex. Yes, No, N/A)
280
(Select box)
Disk Space Used By Call Recording:
• wav49 = 84.5kb
• wav = 833.0kb
• gsm = 85.0kb
• wav49 = 84.0kb
• wav = 827.0kb
• gsm = 84.0kb
• wav49 = 91.0kb
• wav = 863.0kb
• gsm = 91.0kb
• wav49 = 0.38kb
• wav = 3.0kb
• gsm = 0.32kb
Features
Features
• Hide CallerID in OSC:
Whether to hide CallerID in Online Self Care

281
(ex. Yes, No, N/A)
(Option buttons)
• Force "Allow ES CallerID" in Call Forwarding:
Force this option when in the Call Forwarding part of Enhanced Services
(ex. Yes, No, N/A)
(Option buttons)
• Ringtone for Local calls:'
Default Custom Ringtone for Local calls.

(ex. <Simple-3>)
([a-z][0-9])
• Set CallerID for Group Hunt calls:
If numbers set in Group Hunt have one or more remote numbers, those numbers will
receive a different Caller ID on the call. That Caller ID is the one from the original caller
set in its 'Caller ID' enhanced service for the trunk where the call was going through
(ex. Yes, No, N/A)
(Option buttons)
• Only Allow Trunk CallerID within DID range:
If the calls from some other server are going through this system's Multi User
extension, and CallerIDs from other server are matching DIDs on this server, they are
properly shown to callees. Otherwise CallerID is represented as Anonymous
(ex. Yes, No, N/A)
(Option buttons)
• On DID save update ES/CID/Trunks:
Whether to set CallerID on the extension's enhanced services to a value which is the
same as the DID number
(ex. When this option is turned on and you save a DID it will set its number as a
CallerID of the extension to which it points. This will happen when DID is pointing to
said extension, and the CallerID in enhanced services of the extension will be set for
the same trunk that is selected in DID).
(Option buttons)
• Do not allow users sending any CallerID:
282
If enabled, system will not allow extensions/users sending any CallerID
(ex. Yes, No, N/A)
(Option buttons)
Other Networks
The system can be part of the 'default' PBXware MT network where all extensions share
the same unified dial plan. This is achieved by selecting from the select box to which
the PBXware MT network system belongs. Clicking on 'Other network' will open the
following options:
Other Networks
• Mode:
Sets the way other PBXware MT networks are dialed

Example:
• With Access Code - Access code + network number + extension (e.g. *188 8
1000)
• Without Access Code - network number only + extension (e.g. 8 1000)
(Option buttons)
• Name:
Other network name

(ex. Network London)
(Display)
• Number:
Other network access number

(ex. 8)
(Display)
• Trunk:
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Trunk used once the other network number is dialed
(ex. 2554433)
(Display)
• Edit
Edits an other network configuration

(ex. Click to edit an other network configuration)
(Button)
• Delete
Deletes an other network connection

(ex. Click to delete an other network connection from the system)
(Button)
Add/Edit Network
Add/Edit Network
• Name:
Custom Other Network name/identifier

(ex. London FO/7)
([a-z][0-9])
• Prefix:
Number used to access the Other Network (Up to 3 digits allowed)

(ex. If this field is set to '7', dial '*188 7 {NETWORK NUMBER}')
([0-9])
• Strip Prefix:
Should the 'Prefix' number be stripped once dialing the Other network
284
(ex. If the 'Prefix' field is set to '7' and this field is enabled, once user dials *188 7
55510205 the system will dial 55510205. If this field is disabled, 755510205 will be
dialed).
(Option buttons)
• Allowed Range:
Number range allowed to be dialed after the Other Network Access Code
(ex. If this field is set to '2', only extensions on Other Network starting with number 2
will be allowed to dial. If you dial *188 7 1002, our call will fail. But, if we dial *188 7
2000, our call will be transferred to extension 2000).
([a-z][0-9])
• Hidden Prefix:
Prefix number added before dialed number

(ex. If this field is set to 212 and we dialed *188 7 55510205, 21255510205 will be
dialed. This is useful is provider requires a certain code before dialing the destination,
area code inserted automatically etc...)
([a-z][0-9])
• Trunk:
Select trunk that will be used once 'Number' is dialed

(ex. London FO Trunk)
(Select box)
Speed Dial
Speed Dial is used with *130 Access Code. When you dial *130XX, where XX is a two
digit Speed Dial Code, you will dial the extension associated with that code.
The difference here is that you set the Speed Dial Codes for all extensions on the system
not just the ones which have this turned on in ES.
TIP This is useful only if you have more then 6 digits in your extensions.
Speed Dial
285
• Code (XX)
Two digit code which is entered after the Speed Dial Access Code, *130 as default
(ex. 22)
([0-9])
• Speed Dial Name
Short description of the Destination to which this Code points.

(ex. Sales-John)
([a-z][0-9])
• Destination
Destination to which this Code is pointing.

(ex. 1005)
([0-9])
CSV Upload is used when you have all the codes written in simple CSV file in the
following form:
• Code,Name,Destination
CSV Download is used when you want to download already set Dial Codes in a CSV
file
Administration
This section enables remote administration of the system and is set on master tenant.
Administration
• Daemon username:
Daemon username
(ex. admin)
([a-z][0-9])
286
• Daemon passwd:
Daemon password
(ex. GTaXfgtR)
([a-z][0-9])
• Port:
PBXware MT connection port

(ex. 10001)
([0-9])
• AGI Port:
Agi connection port

(ex. 4573)
([0-9])
• Allow LDAP username:
This tells PBXware MT to accept registration from GLOCOM when it is using a

Windows® username for this.
(ex. Yes, No, N/A)
(Option buttons)
Email
System during its operations sends email notifications and alerts to various users and
administrators. These emails can be sent using a built-in 'local mail server' or remote
SMTP server. (Only on master tenant)
Email
• From E-mail:
What email to show in the From: field.

([a-z][0-9])
287
A From E-mail is used to set the From: field in emails that are sent. E-mail
authentication and few additional options can be configured after you go to the Setup
Wizard at https://IPADDRESS:81 and click on SMTP Configuration button.
Email
• E-mail Account:
Address to which email will go if the recipient is not specifically defined

([0-9][a-z] @)
• SMTP Address:
The host to send mail to, in the form "host | IP_addr"

(ex. mail.domain.com)
([0-9][a-z])
• SMTP port:
Port used to send emails to the host

(ex. Default port is 25)
([0-9])
• Authentication:
Select whether authentication with the SMTP server is needed or not

(ex. On)
(Option buttons)
• Username:
Username used for SMTP AUTH

(ex. username)
([a-z][0-9])
288
• Password:
Password used for SMTP AUTH

(ex. password)
([a-z][0-9])
• Allow From Override:
Are users allowed to set their own From: address

Example:
YES - Allow the user to specify their own From: address
NO - Use the system generated From: address
(Option buttons)
• Use SSL:
Specifies whether ssmtp uses TLS to talk to the SMTP server

(ex. The default is NO)
(Option buttons)
• Use Start TLS:
Specifies whether ssmtp does a EHLO/STARTTLS before starting SSL negotiation

(ex. See RFC 2487)
(Option buttons)
Locality
Locality sets from where the system is operating, and it is set for every slave in
particular.
Locality
• Country:
289
Country from which the tenant is operated
(ex. If the tenant is operated from the USA, set USA here)
(Select box)
• Zaptel Zone:
Overrides Automatic Country Detection and this is set on the master tenant where
slaves are using this value.
(ex. It is recommended to keep this setting always set to 'Automatic')
(Select box)
• Indications:
Which indications (Ringing, Busy, etc. sounds) are to be used by the system, also
defined on master tenant.
(ex. If the system is located in USA, set USA here, otherwise select the closest country
to yours)
(Select box)
• Area Code:
Area code from which the tenant is operating

(ex. New York area code - 212)
([0-9])
• National Dialing Code:
Code needed for dialing national destinations

(ex. 1(USA), 0 (United Kingdom and Germany))
([0-9])
• Leave National Code:
(ex. 035123456 will not be striped of 0)
(Option buttons)
Code needed for dialing international destinations

(ex. 011(USA), 00 (United Kingdom and Germany))
290
([0-9])
Emergency Services
When you click on the Emergency Services button, a pop-up window will allow you to
enter Police, Fire, and Ambulance numbers.
Emergency Services
Auto Provisioning
Auto provisioning sets values to be used for the auto provisioning system in order to
create auto provisioning files correctly depending on the UAD locality (local/remote).
This is set on master for all tenants.
Auto Provisioning
• LAN IP:
Local area network IP address used to auto provision local UADs

(ex. 192.168.1.2)
(IP address)
• WAN IP:
Wide area network IP address used to auto provision remote UADs

(ex. 192.168.1.2)
(IP address)
• Use DNS SRV when possible:
Whether Polycom phones should use DNS SRV records to find out SIP port from DNS
server
(ex. Yes, No, N/A)
291
(Option buttons)
Channels
Depending on the CPU power of the server a custom number of channels can be
assigned for various channel types
Channels
• Local Channels:
Total number of all channels used by local UADs

(ex. 12)
([0-9])
• Remote Channels:
Maximum number of active inbound or outbound channels for specific tenant.

(ex. 12)
([0-9])
• Conferences:
Total number of all system conferences

(ex. 8)
([0-9])
• Queues:
Total number of all system ACD queues

(ex. 8)
([0-9])
• Auto Attendants:
Total number of all system IVRs

(ex. 8)
292
([0-9])
• Zaptel:
Total number of all system trunks using ZAPTEL protocol

(ex. 8)
([0-9])
The System will limit the number of channels in order to achieve and maintain excellent
calls and other services' quality
TIP:
Channels can be used as a way of suspending entire tenant without actually having to
delete anything. Setting Local and Remote Channels values to 0 will disable calls on the
Tenant.
Numbering Defaults
Numbering default is set during the initial system set up in order to set how many digits
the system will use as default.
Numbering Defaults
• Number of Digits:
Number of digits used by the system to create local extensions, IVRs, Queues,
Voicemail boxes, Conferences etc.
(ex. This option is available for settings only during the setup wizard install process. In
order to change the number of digits after the setup wizard, please remove all
Extensions, DIDs, Conferences (all apps with network number). The recommended
value for this field is 4).
(Select box)
• PSTN numbering mode:
If this option is turned on, when a number is dialed there is no need for DID, the
system calls that extension automatically.
(ex. Yes, No, N/A)
293
(Option buttons)
• Call groups/Pickup groups:
Select which Call group/Pickup group tenant could use.

(ex. Ctrl+Left Mouse Click to select multiple groups).
(Select box)
Default Codecs
Default codecs can be set for the following groups:
• Local - Local extensions

• Remote - Remote extensions and Trunks
• Network - PBXware MT network(two or more servers)
Tip
Once a local/remote extension or a network is added/edited, only the codecs allowed

here will be available for the extension/network usage.
Default Codecs
Available Codecs:
• ITU G.711 ulaw' - 64 Kbps, sample-based, used in US

• ITU G.726 - 16/24/32/40 Kbps
294
• LPC10 - 2.5 Kbps
Monitoring
Monitoring sets alarms and notifications at which the system will monitor itself for
normal operation and where by appropriate notifications are sent if alarms are triggered.
Tip
Reloading the system will not interrupt any services while restarting the system does
stop and starts all system services. This is set on master tenant and affects all other
tenants.
Monitoring
• Monitor (mins):
Time interval at which the system should check if Asterisk is down. If down, the system
will try to start it and will send a notification email about the stop/start action
(ex. 15)
([0-9])
• Reload Type:
Select whether to reload the system at some specific time of a day or in regular time
intervals (hourly)
295
(ex. Setting this option to 'Time of the day' and 'Reload (hours)' = '2' will reload the
system every day at 02:00 hours. Setting this option to 'Regular Interval' and 'Reload
(hours)' = '2' will reload the system every two hours)
(Select box)
• Reload (hours):
This field is active only when the 'Reload Type' option is selected
(ex. Setting 'Reload Type' = 'Time of the day' and this option to '2' will reload the
system every day at 02:00 hours. Setting 'Reload Type' = 'Regular Interval' and this
option to '2' will reload the system every two hours)
(Select box)
• Restart Type:
Select whether to restart the system at some specific time of day or in regular time
intervals (hourly)
(ex. Setting this option to 'Time of the day' and 'Restart (hours)' = '2' will restart the
system every day at 02:00 hours. Setting this option to 'Regular Interval' and 'Restart
(hours)' = '2' will restart the system every two hours)
(Select box)
• Reastart (hours):
This field is active only when the 'Restart Type' option is selected
(ex. Setting 'Restart Type' = 'Time of the day' and this option to '2' will restart the
system every day at 02:00 hours. Setting 'Restart Type' = 'Regular Interval' and this
option to '2' will restart the system every two hours)
(Select box)
• Notification e-mail:
Email address on which reload/restart notification is sent

([a-z][0-9])
DynDNS Update Service
296
DynDNS Update Service
• Run every:
Select the time interval at which the DynDNS update request will be sent
(ex. Select 'minute' to send request to the DynDNS server every minute)
(Select box)
• Mode:
Set the way the DynDNS service will work on the system
(ex. 'Update & Monitor' will monitor and update/reload the IP address(es) of
monitored servers once they change. 'Monitor only' will monitor for IP address change
but will not update/reload)
(Select box)
• Dynamic Hostname:
Hostname to which requests are periodically sent

(ex. domain.dyndns.org)
([a-z][0-9])
• Username:
Username for DynDNS authentication

(ex. Type the username used for DynDNS authentication here)
([a-z][0-9])
• Password:
Password for DynDNS authentication

(ex. Type the password used for DynDNS authentication here)
([a-z][0-9])
297
• DynDNS Server:
DynDNS server name

(ex. members.dyndns.org, (default value))
([a-z][0-9])
• CheckIP Server:
DynDNS server for checking the IP addresses

(ex. checkip.dyndns.org, (default value))
([a-z][0-9])
• Monitor Servers:
Dyndns domains of monitored servers separated by a blank space

(ex. serverone.dyndnd.org)
([a-z][0-9])
Tenant packages
Tenant packages are "presets" that are used when determining how many of the
following, slaves can have: extensions, voicemails, queues, IVRs, conferences and ring
groups.
Tenant packages
• Package name
Name of the package.

(ex. Full package)
(Display)
• Usage count
Number of tenants using this package.

(ex. 4)
298
(Display)
• Edit
Edits the package configuration

(ex. Click to edit package configuration)
(Button)
• Delete
Deletes the package

(ex. Click to delete the package)
(Button)
Add Package
Add Package
• Name:
Name of the tenant package.

(ex. Medium package)
([0-9][a-z])
• Extensions:
Number of extensions that can be created on a tenant using this package.

(ex. 10)
([0-9])
• Voicemails:
Number of voicemails that can be created on a tenant using this package.
299
(ex. 6)
([0-9])
• Queues:
Number of queues that can be created on a tenant using this package.

(ex. 2)
([0-9])
• IVRs:
Number of IVRs that can be created on a tenant using this package.

(ex. 3)
([0-9])
• Conferences:
Number of conferences that can be created on a tenant using this package.

(ex. 4)
([0-9])
• Ring Groups:
Number of ring groups that can be created on a tenant using this package.
(ex. 2)
([0-9])
Protocols
Protocol is a set of rules that allows UAD, systems, etc. to communicate using a set
standard.
Supported protocols are:
• SIP
• IAX
• WOOMERA
SIP
300
SIP (Session Initiated Protocol, or Session Initiation Protocol), is a signaling protocol
for Internet conferencing, telephony, presence, events notification, and instant
messaging. The protocol initiates call setup, routing, authentication, and other feature
messages to end points within an IP domain.
General
General
• Port:
SIP bind port

(ex. 5060, (default))
([0-9])
• Bind Address:
SIP bind IP address

(ex. 0.0.0.0, (default))
([0-9])
• SRR lookup:
Enable DNS SRV lookups on outbound calls

(ex. Disabling this option will disable SIP calls based on domain names between SIP
users on the Internet)
(Option buttons)
• Qualify:
Timing interval in milliseconds at which a 'ping' is sent to a host in order to find out its
status
301
(ex. Set this field to 2000, for example. If more time than provided here is needed to
reach the host, the host is considered offline)
([0-9])
• Context:
Default context for incoming calls

(ex. For security reasons it is recommended to keep this field set at 'invalid-context')
([a-z][0-9])
• Language:
Default language settings for all users/peers

Example:
Set this option to 'en' (English), for example
([a-z])
• Music on Hold:
Set the default MOH (Music on Hold) class for all SIP calls
(ex. Set 'default', for example, to play 'default' MOH class to all SIP calls when placed
on hold)
(Select box)
• T.38 Passthru Fax support:
Enables T.38 fax pass-through on SIP to SIP calls, provided both parties have T38
support enabled.
(ex. Yes, No, N/A)
(Option buttions)
• Allow Overlap Dialling:
Allow overlap dialing.

(ex. Yes, No, N/A)
(Option buttons)
• Allow Transfer:
Disable all transfers (unless enabled in peers or users). Default is enabled.

(Yes, No, N/A)
302
(Option buttons)
NAT
NAT
• External IP:
External IP/Public/Internet address system uses

(ex. If your system is behind NAT, set this option to Public/Internet IP address system
uses when registering with other proxies over Internet)
([0-9])
• External Host (DynDNS):
DynDNS address system uses

(ex. If your system is behind NAT, along with the External IP address you may use the
DynDNS service as well. Set this field to DynDNS host)
([0-9])
• External Host Refresh:
How often to refresh External DynDNS host (if used)

(ex. Time in seconds (eg. 10))
([0-9])
• Match External IP Locally:
Only substitute the externip or externhost setting if it matches your localnet setting.
(ex. Yes, No, N/A)
(Option buttons)
• Local network:
If system is used in local network, set the local network address here
303
(ex. 192.168.0.0/255.255.0.0)
([0-9])
• NAT:
Global SIP NAT setting which affects all users/peers

(ex. Set this option to 'Yes' if system is behind NAT)
(Option buttons)
Security
Security
• Always Reject with 401:
(Option buttons)
• Allow guest:
(Option buttons)
• Allow External INVITEs:
(Option buttons)
• Trust Remote-Party-ID:
(Option buttons)
• Allow REDIR:
Option buttons
304
RTP
RTP
• RTP timeout:
Max RTP timeout

Example:
All calls (if not on hold) will be terminated if there is no RTP activity for the number of
seconds set here (60, for example)
([0-9])
• RTP hold timeout:
Max RTP hold timeout. NOTE: This field must be a higher number than set under 'RTP
timeout'
Example:
All calls on hold will be terminated if there is no RTP activity for the number of seconds
set here (300, for example)
([0-9])
• RTP keer-alive:
Send keep-alives in the RTP stream to keep NAT open (default is off - zero).
([0-9])
DTMF
DTMF
• DTMF Mode:
Set the default DTMF mode

(ex. rfc2833)
305
(Select box)
• Relax DTMF:
Relax DTMF handling

(ex. Set this field to 'Yes' if having problems with DTMF modes)
(Option buttons)
Misc
Misc
• Pedantic checking:
Enable slow, pedantic checking for Pingtel and multi-line formatted headers for strict
SIP compatibility
(ex. It is recommended to set this field to 'No')
(Option buttons)
• Generate inband ringing:
Set whether the system generates in-bank ringing

(ex. You're recommended to set this option to 'Never')
(Select box)
• Video support:
Set whether the system generates in-bank ringing

(ex. You're recommended to set this option to 'No')
(Option buttons)
• Max Call Bitrate:
Maximum bitrate for video calls (default 384 kb/s).
306
(ex. 512)
([0-9])
• Send Remote-Party-ID:
Should 'Remote-Party-ID' be added to uri

(ex. You're recommended to set this option to 'No' unless required otherwise)
(Option buttons)
• Add ;user=phone:
Should ';user=phone' be added ot uri

(Option buttons)
• Compact Headers:
Should compact SIP headers be sent

(Option buttons)
• Manager events on SIP events:
Should manager events be generated if SIP UAD/Phone performs some event (Hold for
example)
(Option buttons)
Authentication
Authentication
• User Agent:
Set the 'User Agent' string

Example:
307
'Custom string', for example
([a-z][0-9])
• Realm:
Realm for digest authentication

Example:
'Custom string', for example
([a-z][0-9])
• Auth debugging:
Should authentication be debugged

(ex. Setting this option to 'Yes' will increase the amount of debugging traffic)
(Option buttons)
Registration
Registration
• Length of i/o reg:
([0-9])
• Def. Length of i/o reg:
([0-9])
• Registration context:
Should the system dynamically create and destroy noop priority 1 extension for a peer
who (un)registers with us
(ex. sipregistrations)
([a-z][0-9])
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• Registration timeout:
Number of seconds after which registration times out

(ex. Default value 20)
([0-9])
• Register attempts:
Number of registration attempts

(ex. One 'Register timeout' equals one 'Registration attempts'. Default value 10)
([0-9])
• Minimum roundtrip time for monitored hosts:
Minimum roundtrip time for messages to monitored hosts. Defaults to 100 ms.
(ex. 300)
([0-9])
MWI
MWI
• MWI Mime-type:
Allow overriding of mime type

(ex. Default value 'text/plain')
([a-z])
• Check MWI time:
Default time between mailbox checks for peers

(ex. Default value 10)
([0-9])
• Voicemail extension:
309
Dialplan extension to reach mailbox. This option sets the 'Message-Account' in the
MWI notify message
(Default value 'asterisk')
([a-z][0-9])
Subscriptions
Subscriptions
• Allow Subscriptions:
Disable support for subscriptions. Default is Yes.

(ex. Yes, No, N/A)
(Option buttons)
• Subscribe Context:
Set a specific context for SUBSCRIBE requests (Useful to limit subscriptions to local
extensions)
([a-z][0-9])
• Notify on RINGING:
Notify subscriptions on RINGING state

(ex. Yes, No, N/A)
(Option buttons)
• Notify on HOLD:
Disable support for subscriptions. Default is No.

(ex. Yes, No, N/A)
(Option buttons)
310
Domains
Domains
• Domain:
Set the default domain for this host

(ex. If configured, Asterisk will only allow INVITE and REFER to non-local domains. Use
'sip show domains' to list local domains)
([a-z][0-9])
• Auto Domain:
Turn this on to have Asterisk add a local host name and local IP to the domain list.
(ex. If the system host name is set to 'my_system', with this feature set to 'On',
'my_system' will be automatically added to the domain list)
(Option buttons)
• From Domain:
Change the 'From:' headers

(ex. Keep this field empty unless requested otherwise)
([a-z][0-9])
• Allow External Domains:
Should domains not serviced by this server be (dis)allowed

(Option buttons)
Codecs
311
Codecs
• Disallow:
Set the codecs that are not allowed for usage

to 'Disallow All' and it cannot be modified)
(Read only)
• Allow:
Set the codecs that are allowed for usage

(ex. Only the codecs set under 'Settings: Server' will be available to choose from)
(Check box)
• Auto-Framing (RTP Packetization):
If autoframing is turned on, system will choose the packetization level based on the
remote ends preferences.
(ex. Yes)
(Option button)
• Non-standard G726:
If the peer negotiates G726-32 audio, use AAL2 packing order instead of RFC3551
packing order.
(ex. Yes, No, N/A)
(Option buttons)
Available Codecs:

• ITU G.726 - 16/24/32/40 Kbps
• LPC10 - 2.5 Kbps
312
Jitter Buffer
Jitter Butter
• Enable Jitterbuffer:
Enables the use of a jitterbuffer on the receiving side of a SIP channel. An enabled
jitterbuffer will be used only if the sending side can create and the receiving side can
not accept jitter. The SIP channel can accept jitter, thus a jitterbuffer on the receive SIP
side will be used only if it is forced and enabled.
(ex. Yes, No, N/A)
(Option buttons)
• Force Jitter Buffer:
Should we force jitter buffer (default value 10)

(ex. Jitter buffer is usually handled by the UADs/Phones. But in case if these do this
poorly jitter buffer can be enforced on PBXware MT side)
([0-9])
• Max length (ms):
Max length of the jitterbuffer in milliseconds.

(ex. 300)
([0-9])
• Re-sync threshold:
Resync the threshold for noticing a change in delay measured

(ex. 1000)
([0-9])
• Implementation:
313
Jitterbuffer implementation used on a receiving side of a SIP channel. Defaults to
"fixed".
(ex. adaptive)
(Select box)
• Logging:
Enables jitterbuffer frame logging. Defaults to "no".

(ex. Yes, No, N/A)
(Option buttons)
SIP Debugging
Adjust options that affect SIP debugging on the system
SIP Debugging
• SIP Debug:
Should SIP Debug be turned on all the time

(Option buttons)
• Record History:
Should SIP history be recorded

Example:
Select 'Yes' to record SIP history. Example history information:
* SIP Call
1. TxReqRel INVITE / 102 INVITE
2. Rx SIP/2.0 / 102 INVITE /100 Trying
3. CancelDestroy
4. Rx SIP/2.0 / 102 INVITE /180 Ringing
5. CancelDestroy
6. Rx SIP/2.0 / 102 INVITE /200 OK
7. CancelDestroy
314
8. Unhold SIP/2.0
9. TxReq ACK / 102 ACK
10. TxReqRel INVITE / 103 INVITE
11. Rx SIP/2.0 / 103 INVITE /200 OK
12. CancelDestroy
13. Unhold SIP/2.0
14. TxReq ACK / 103 ACK
(Option buttons)
• Dump History:
Dump SIP history at end of SIP dialogue. SIP history is output to the DEBUG logging
channel.
(ex. Yes, No, N/A)
(Option buttons)
Additional config
This option is used for providing additional config parameters for SIP configuration
files. Values provided here will be written into these configuration files.
Additional config
IAX
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IAX (Inter asterisk exchange) is a simple, low overhead and low bandwidth VoIP
protocol designed to allow multiple PBXware MT to communicate with one another
without the overhead of more complex protocols.
General
General
• Port:
iIAX bind port

(ex. 4569, (default))
([0-9])
• Bind Address:
IAX bind IP address

(ex. 0.0.0.0, (default))
([0-9])
• IAX compatible:
Should layered switches or some other scenario be used

(ex. Set to yes if you plan to use layered switches or some other scenario which may
cause some delay when doing a lookup in the dialplan)
(Select box)
• Language:
Default language settings for all users/peers

Example:
Set this option to 'en' (English) for example
([a-z])
316
• Bandwidth:
Set the bandwidth to control which codecs are used in general

(ex. Select between low, mid or high)
(Select box)
• Maximum number of IAX helper threads:
Establishes the number of extra dynamic threads that may be spawned to handle I/O.
(ex. 150)
([0-9])
• Allow IAXy firmware download:
Controls whether this host will serve out firmware to IAX clients which request it.
(ex. Yes, No, N/A)
(Option buttons)
Jitterbuffer
Jitterbuffer
• Jitter Buffer:
Turn off jitter buffer for this peer

(ex. Yes, No, N/A)
(Option buttons)
• Force Jitter Buffer:
Should we force jitter buffer (default value 10)

(ex. Jitter buffer is usually handled by the UADs/Phones. But in case if these do this
poorly jitter buffer can be enforced on PBXware MT side)
([0-9])
317
• Max. jitterbuffer interpolations:
The maximum number of interpolation frames the jitterbuffer should return in a row
(ex. 1000)
([0-9])
• Max. Jitter buffer:
A maximum size for the jitter buffer. Setting a reasonable maximum here will prevent
the call delay from rising to silly values in extreme situations; you'll hear SOMETHING,
even though it will be jittery.
(ex. 1000)
([0-9])
• Resync Treshold:
Resync the threshold for noticing a change in delay measured

(ex. 1000)
([0-9])
Billing
Billing
• AMA Flags:
These flags are used in the generation of call detail records (e.g 'default')
(ex. Select between 'default', 'omit', 'billing' or 'documentation')
(Select box)
• Account code:
Default account for CDRs (Call Detail Records)

(ex. lars101)
([a-z][0-9])
318
Authorization
Authorization
• Auth debugging:
Should authentication be debugged

(ex. Setting this option to 'Yes' will increase the amount of debugging traffic)
(Option buttons)
• Max Auth requests:
Maximum number of outstanding authentication requests waiting for replies. Any

further authentication attempts will be blocked
(ex. 10)
([0-9])
• Delay Reject:
Set this option to 'Yes' for increased security against brute force password attacks
(ex. Yes)
([0-9])
Registration
Registration
• Registration context:
If specified PBXware MT will dynamically create and destroy a NoOp priority 1

extension for a given peer who registers or unregisters with us
(ex. iaxregistration)
([a-z][0-9])
319
• Min Registration Expire:
Minimum amounts of time that IAX peers can request as a registration expiration
interval (in seconds).
(ex. 60)
([0-9])
• Max Registration Expire:
Maximum amounts of time that IAX peers can request as a registration expiration
interval (in seconds).
(ex. 60)
([0-9])
Trunk
Trunk
• Trunk frequency:
How frequently to send trunk msgs (in ms)

(ex. 20)
([0-9])
• Trunk Timestamps:
Should we send timestamps for the individual sub-frames within trunk frames
(ex. Yes)
(Option buttons)
Misc
Misc
320
• Disable UDP checksums:
Should checkums will be calculated

(ex. Yes)
(Option buttons)
• Auto-kill:
If no response is received within 2000ms, and this option set to yes, cancel the whole
thing
(ex. Yes)
(Option buttons)
Codecs
Codecs
• Codec Priority:
This option controls the codec negotiation of an inbound IAX calls.

Example:
• caller - Consider the callers preferred order ahead of the host's.

• host - Consider the host's preferred order ahead of the caller's.
• disabled - Disable the consideration of codec preference altogether (this is the
original behaviour before preferences were added)
• reqonly - Same as disabled, only do not consider capabilities if the requested
format is not available the call will only be accepted if the requested form
(Read only)
• Disallow:
Set the codecs extension is now allowed to use

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(Read only)
• Allow:
Set the codecs extension is allowed to use

(Check box)
Available Codecs:

• ITU G.726 - 16/24/32/40 Kbps
• LPC10 - 2.5 Kbps
Additional config
This option is used for providing additional config parameters for IAX configuration
files.
Values provided here will be written into these configuration files.
Additional config
WOOMERA
322
WOOMERA is the protocol which is used ba Sangoma cards.
WOOMERA
• WOOMERA Server:
This is the remote IP address of the Woomera Server

(ex. 192.168.0.13)
([0-9])
• WOOMERA Server Port:
This is the remote port number of the Woomera Server

(ex. 42420)
([0-9])
• WOOMERA Audio Server:
IP address of remote SMG Audio Server

(ex. 192.168.0.13)
([0-9])
• Default Context:
Incoming Context for Woomera Channels. All incoming calls will be forwared to
defined context in extensions.conf
(ex. dids)
([a-z][0-9])
• Debug:
Debug flag is used to enable/disable woomera channel debugging level.

(ex. 2)
323
([0-9])
• Enable Incoming DTMF:
This option will enable Rx DTMF detection on each channel. The Tx DTMF is
automatically enabled on SMG
(ex. Yes, No, N/A)
(Option buttons)
• Enable JitterBuffer:
Enable Jitterbuffer for this profile

(ex. Yes, No, N/A)
(Option buttons)
• Enable Progress Messaging:
Enable Asterisk Progress Messaging used for Asterisk Early Audio with ZAP and SIP
(ex. Yes, No, N/A)
(Option buttons)
• Coding:
Coding is necessary to identify ulaw/alaw

(ex. ulaw)
(Select box)
• RX Gain:
Set RX gain
(ex. 2)
([0.0 - x.0])
• TX Gain:
Set TX gain
(ex. 2)
([0.0 - x.0])
• Language:
Set preffered language

(ex. en)
324
([a-z])
Providers
PBXware MT comes with a range of pre-configured VoIP and PSTN service providers
in order to allow an easy way of adding trunks into the system. This screen allows for
the addition of custom providers by clicking on 'Add Custom Provider'.
In addition, 'Import Providers' allows for the update of currently pre-configured service
providers.
Providers
• Provider:
Provider name
(ex. Generic Analog)
(Display)
• Protocol:
Protocol provider uses

(ex. zaptel)
(Display)
• Type:
Service type
(ex. pstn/voip)
(Display)
• Edit
Edit Provider configuration

(ex. Click to edit Provider configuration)
325
(Button)
• Delete
Delete Provider configuration

(ex. Click to delete Provider configuration from the system
(Button)
PSTN
From here you can select two PSTN types of providers: ZAPTEL (analog and PRI),
MISDN (BRI).
ZAPTEL
PSTN
• Type:
Service type
(ex. pstn/voip)
(Select box)
• Protocol:
Protocol provider uses

(zaptel/capi)
(Select box)
• Provider:
326
Provider name
(ex. BT)
([a-z][0-9])
• Country:

(ex. USA)
(Select box)
• E164 Accepted:
Does the Provider support dialing destinations in E164 format

(Option buttons)
National dialing code at the Provider destination

(ex. For USA, 1, United Kingdom and Germany 0)
([0-9])
national code will not be stripped from the dialed number. NOTE: Before setting this
option to 'Yes', go to 'Settings: Servers' and enable this option as well.
([0-9])
International dialing code at the Provider destination

(ex. For USA, 011, United Kingdom and Germany, 00)
([0-9])
• Local Area Code:
Add local area code to dialed number, if required by the service provider. (By default,
the local area code is stripped when dialing)
327
(ex. User dials 55510205, the local area code is 212. If the call goes through this trunk
PBXware MT will dial 21210205)
([0-9])
• Write dialing code:
Should National and International prefix be written into configuration files

(ex. Enable this option if required by the provider)
([0-9])
• Zaptel Devices:
Select which zaptel device the system is to use

Example:
• X100P
• TDM400P
• TDM10B
• TDM20B
• TDM30B
• TDM40B
• TDM11B
• TDM12B
• TDM13B
• TDM21B
• TDM22B
• TDM23B
• T100P
• E100P
• TE110P
• TE410P
• TE405P
• hfcISDN
• quadBRI
• A101u
• A102u
• A104u
(Option buttons)
Tip
Please configure Zaptel devices after you save the provider by clicking the 'Configure'
button next to a selected device.
328
MISDN
MISDN
• Country:
Country which will be shown as a default when adding a trunk with the current
provider.
(ex. United Kingdom)
(Select box)
• National dialing code:
Prefix when dialing national numbers.

(ex. 0)
([0-9])
• International dialing code:
Prefix used when dialing international numbers.

(ex. 00)
([0-9])
• E164 Accepted:
Does the provider used support dialing destinations in E164 format

COUNTRY_CODE + AREA_CODE + DIALED_NUMBER. For example, if you dial 55510205,
(Option buttons)
• Pass-thru mode:
Pass the digits dialed without any conversion (E164, National, Area code). NOTE: When
active, 'Leave National Code and 'Local Area Code' will be disabled
(ex. If this option is disabled, PBXware MT will convert all dialed numbers to E164
format (COUNTRY_CODE + AREA_CODE + DIALED_NUMBER) and then make a call to
329
the converted number. If this option is enabled, PBXware MT will call directly
DIALED_NUMBER without making any number conversions)
(Option buttons)
option to 'Yes', go to 'Settings: Servers' and enable this options as well.
(ex. 035123456 will not be stripped of 0)
(Option buttons)
(ex. User dials 55510205, local area code is 212. If call goes through this trunk,
([0-9])
• Incoming Limit:
Number of simultaneous incoming calls the trunk can handle

(ex. 4 equals four simultaneous incoming calls. Any additional calls will get the busy
sound)
([0-9])
• Outgoing Limit:
Number of simultaneous outgoing calls the trunk can handle

(ex. 4 equals four simultaneous outgoing calls. Any additional calls attempting to use
this trunk will be rejected or will be redirected to other trunks depending on what is
set in the system/extensions)
([0-9])
Send an e-mail when the outgoing limit is reached

(ex. Yes, No, N/A)
(Option buttons)
• mISDN Devices:
330
Select the mISDN device that the system is to use
Example:
• B410P
• USB-ISDN
(Option buttons)
VOIP
This is where you select which type of VoIP you are going to create a custom provider
for.
SIP
SIP
• User Type:
User's relationship to the system

Example:
• user - Provider accepts incoming calls only

• peer - Provider makes outgoing calls only
• friend - Provider does both incoming and outgoing calls
(Select box)
• DTMF Mode (Dual Tone Multi-Frequency):
DTMF mode used by provider. A specific frequency (consisting of two separate tones)
to each key so that it can easily be identified by a microprocessor
Example:
• inband - inband audio(requires 64 kbit codec - alaw, ulaw)

331
(Select box)
• Country:

(ex. USA)
(Select box)
National dialing code used at the Provider destination

([0-9])
International dialing code used at the provider destination

([0-9])
• E164 Accepted:
Does the trunk support dialing destinations in the E164 format

(ex. Enabling this option will reformat any dialed number into following form
(Option buttons)
• Pass-thru Mode:
If this option is enabled, the number which is dialed is passed through the trunk
without modification
(ex. Yes, No, N/A)
(Option buttons)
332
([0-9])
(ex. User dials 55510205, local area code is 212. If the call goes through this provider,
([0-9])
• Canreinvite:
Should you allow RTP voice traffic to bypass Asterisk. NOTE: All enhanced services for
the extension have to be disabled
(ex. Some devices do not support this especially if one of them is behind a NAT)
(Options buttons / Select box)
• Default IP:
IP address to be used until registration

(ex. 192.168.1.1)
(IP Address)
• Incoming Limit:
Number of simultaneous incoming calls the Provider can handle

sound)
([0-9])
• Outgoing Limit:
Number of simultaneous outgoing calls the Provider can handle

this Provider will be rejected or will be redirected to other Providers depending on
what is set in the system/extensions)
([0-9])

(ex. Yes, No, N/A)
333
(Option buttons)
• Host:
Provider IP address
(ex. Enter host IP, 192.168.1.1 for example or set 'dynamic' if host is behind dynamic IP
address)
([0-9][a-z])
• Peer Host:
IP of a peer host that the system sends the calls to

(ex. 192.1168.1.1)
(IP Address)
• Auth:
Global credentials for outbound calls, i.e. when a proxy challenges your PBXware MT
for authentication these credentials override any credentials in peer/register definition
if realm is matched.
(ex. john:dfgERG@pbxware)
([a-z][0-9])
• Register:
Method for registering to remote server

(ex. Providers may require different way of registration to their server. You may
choose between 'registration not required', 'register with phone number' and 'register
with username')
(Select box)
• Register suffix:
Service provider may request different registration methods for their services. Select
the proper method, as required by the provider
(ex. 1234567)
([0-9])
• Insecure:
Which of the selected options are not used for authentication.

(ex. very - Do not require authentication of incoming INVITEs)
334
(Select box)
• Disallow:
This field is very unique. In order to work properly this setting is automatically set to
'Disallow All' and cannot be modified.
(ex. all)
(Display)
• Allow:
Codecs that are allowed in 'Settings: Server' will be enabled for selection.
(Check boxes)
• Auto-Framing
Automatically adjust RTP packet size with the registered device.

(ex. Yes)
(Option buttons)
Available Codecs:

• ITU G.722
• 'ITU G.723.1 - 5.3/6.3 Kbps, 30ms frame size
• ITU G.726 - 16/24/32/40 Kbps
• ITU G.726 AAL2
• LPC10 - 2.5 Kbps
• H.264 Video
IAX
335
IAX
• User Type:
User's relationship to the system

Example:
• user - Provider accepts incoming calls only

• peer - Provider makes outgoing calls only
• friend - Provider does both incoming and outgoing calls
(Select box)
DTMF mode used by provider. A specific frequency (consisting of two separate tones)
to each key so that it can easily be identified by a microprocessor
Example:
• inband - inband audio(requires 64 kbit codec - alaw, ulaw)

(Select box)
• Country:

(ex. USA)
(Select box)

([0-9])
336

([0-9])
• E164 Accepted:

(Option buttons)
• Pass-thru Mode:
(ex. Yes, No, N/A)
(Option buttons)
option to 'Yes', go to 'Settings: Servers' and enable this option as well.
([0-9])
(ex. User dials 55510205, the local area code is 212. If the call goes through this
provider, PBXware MT will dial 21210205)
([0-9])
• Canreinvite:
Should you allow RTP voice traffic to bypass Asterisk. NOTE: All enhanced services for
the extension have to be disabled
(ex. Some devices do not support this, especially if one of them is behind a NAT)
337
(Options buttons / Select box)
• Default IP:
IP address to be used until registration

(ex. 192.168.1.1)
(IP Address)
• Incoming Limit:
Number of simultaneous incoming calls Provider can handle

sound)
([0-9])
• Outgoing Limit:
Number of simultaneous outgoing calls that a Provider can handle

([0-9])

(ex. Yes, No, N/A)
(Option buttons)
• Notransfer:
Disable the native IAX transfer

(Option buttons)
• Send ANI:
Should ANI ("super" Caller ID) be sent over this Provider

(ex. Set 'Yes' to enable)
(Option buttons)
• Trunk:
Use IAX2 trunking with this host
338
(Option buttons)
WOOMERA
WOOMERA
• Country:

(ex. USA)
(Select box)

([0-9])

([0-9])
• E164 Accepted:

(Option buttons)
• Pass-thru Mode:
339
(ex. Yes, No, N/A)
(Option buttons)
([0-9])
(ex. User dials 55510205, the local area code is 212. If the call goes through this
provider, PBXware MT will dial 21210205)
([0-9])
• Incoming Limit:
Number of simultaneous incoming calls that the Provider can handle

sound)
([0-9])
• Outgoing Limit:
Number of simultaneous outgoing calls the Provider can handle

([0-9])

(ex. Yes, No, N/A)
(Option buttons)
340
E-mail Templates
Here you can decide how the mails that are sent by the system should look. By default
several macros are used and if you want to modify the template using them, they are
pretty self-explanatory.
New Extension Templates
Set e-mail templates which are used to send basic information on newly created
extensions.
New Extension Templates
Billing E-mail Templates
This is used to set E-mail template for extensions Soft and Hard billing limits.
341
Billing E-mail Templates
Incoming/Outgoing Limit Templates
Set the e-mail template used for sending emails when an extension reaches its limit for
incoming/outgoing number of channels.
342
Set the e-mail template used for sending emails when Fax-to-email is set on a DID.
Voicemail
Calls are diverted to Voicemail when a user is unavailable or has the phone powered
off, or when a call is transferred to voicemail by user. The phone alerts the user to
indicate the receipt of a message.
Once the user is transferred to the party's voice box a 'Please leave a detail message
after the tone. If you would like to speak to the operator, press 0' message will be heard.
The user has two options:
1. To leave a voice message that is ended by pressing the # key or by hanging up

2. To reach an operator by dialing 0
If 0 is dialed a 'Press 1 to accept this recording, otherwise please continue to hold'

message will be heard. The user has two options:
1. Press 1 to save your message and dial the operator. 'Please hold while I try that
extension' message played.
2. Continue to hold to delete your message and dial the operator. 'Message deleted,
please hold while I try that extension' message played.
343
General Voicemail
General fields are most required by voicemail
General Voicemail
• Format:
Audio format in which voice messages are recorded

(ex. If 'wav49' is selected here, all voice messages will be saved in this format. See
below for disk usage)
(Select box)
• Max Message Length:
Maximum length of a voice message in seconds

(ex. By default this field is set to '180' seconds (3 minutes))
([0-9])
• Min Message Length:
Minimum length of a voice message in seconds

(ex. Default value set to '3' seconds. Messages that last less are discarded)
([0-9])
• Max Greeting Length:
Maximum length in seconds of the user recorded voicemail greeting message

(ex. Default values set to '60' seconds)
([0-9])
• Max Seconds of Silence:
344
Maximum length of silence in a voice message in seconds
(ex. Default value set to '10' seconds. Silence longer than set here will end a voice
message)
([0-9])
• Silence Threshold:
Silence detection threshold

(ex. Default value set to '128'. The higher the number, the more background noise is
added)
([0-9])
Delay a number of seconds before asking the user for the 'Password'
(ex. If you hear a partial sound file played asking the user for a password, set '1' or '2'
here to add a second or two of silence before the sound file is played)
([0-9])
• Max Files per Directory:
Maximum number of voicemail messages per voicemail directory

(ex. Each voice box has following directories (INBOX, Old, Work, Family, Friends, Cust1,
Cust2, Cust3, Cust4, Cust5). Set this field to '100' to allow a 100 voice messages per
each voice directory)
([0-9])
• Min PIN Length:
Defines the minimum length of PIN used for Voicemail

(ex. 4)
([0-9])
• wav49 = 91.0kb
• wav = 863.0kb
• gsm = 91.0kb
345
• wav49 = 0.38kb
• wav = 3.0kb
• gsm = 0.32kb
E-mail Settings
Customize the display of emails that notify the user of new voicemail messages.
E-mail Settings
• Server E-mail:
This email address is used to identify from whom the email came
(ex. If this field is set to '[email protected]' in the email header the following line is
added '"$FROM": string <[email protected]>')
([a-z][0-9])
• Send Attachment:
Send the voice message as an attachment to the user's email.

(Option buttons)
Delete a voice message after sending it as an attachment to a user email.

be deleted from the voice box after it has been emailed to B).
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(Option buttons)
• Skip[PBX]: in subject:
Should 'PBX' be skipped from the voicemail title

(ex. If set to 'No' - 'Subject: [PBX']: New message M in mailbox B' will be displayed in
email subject line)
(Option buttons)
• "From:" string:
Override a portion of the From: line in the voicemail notification email

(ex. If this option is set to 'MARS', then the email will have one additional line in its
header 'From: MARS')
([a-z][0-9])
• E-mail Subject:
Customize the voicemail notification email subject

(ex. Use custom variables '${VM_MSGNUM}' for message number and
'${VM_MAILBOX}' to create a custom email subject. 'PBXware MT: New message
${VM_MSGNUM} in mailbox ${VM_MAILBOX}' for example. Email subject would look
like this 'PBXware MT: New message 3 in mailbox 1002')
([a-z][0-9])
• E-mail body:
Customize voicemail notification email body

Example:
Use custom variables '${VM_NAME}' for User name, '${VM_MAILBOX}' for mailbox
number, ${VM_DATE} for voicemail date and ${VM_DUR} for voicemail duration to
create custom email body. For example:
Dear, ${VM_NAME}\n\nYou have a new voicemail message on:\n\nmailbox:

${VM_MAILBOX}\n\nleft at: ${VM_DATE}\n\n${VM_DUR} long.
([a-z][0-9])
• Charset:
Select which charset to use when sending mails.

(ex. ISO-8859-1)
([a-z][0-9])
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• Mail command:
Overrides the default mailer command

(ex. Default value '/usr/sbin/sendmail -t')
([a-z][0-9])
Run Application
Run custom applications on certain voicemail actions
Ran Application
• On Voicemail:
Run a custom application when a new voicemail is received

Example:
Set this field to
'/usr/bin/myapp'
to execute
'myapp'
application when a new voicemail arrives
(([a-z][0-9])
• 'On Password Change:
Run custom application when a voicemail password is changed

Example:
Set this field to
'/usr/bin/myapp'
to execute the
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'myapp'
application when the voicemail password changes)
([a-z][0-9])
Main Voicemail Menu
Edit settings for the main voicemail menu.
Main Voicemail Menu
recorded.
(ex. Once the voice box is checked for new messages, if this option is set to 'Yes',
'Received at {$DATE}. The from phone number {$NUMBER}' will be played, giving more
details about the message originator).
(Option buttons)
• Say Caller ID:
(ex. If this option is set to 'Yes', when checking voicemail, the 'From phone number
(Option buttons)
• Skip ms on playback:
Interval in milliseconds to use when skipping forward or reverse while a voicemail

message is being played
(ex. If this field is set to '3000', when listening to voice message skip 3 seconds on
rewind/fast forward)
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([0-9])
• Max login attempts:
Maximum number of login retries before user gets disconnected

(ex. By default this field is set to '3'. After 3 unsuccessful login attempts user gets
disconnected)
([0-9])
• On Delete, play next message:
After a voice message has been deleted, should the system automatically play the next
message from voice inbox
(ex. Select 'Yes' to automatically playback the next voice message after you've deleted
the old one)
(Option buttons)
Advanced Features
Advanced Features
box.
Example:

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(Option buttons)
• Allow Operator:

Example:
following options):
(Option buttons)
• System Operator:

([0-9])
• Send Voicemail:
Change context to send voicemail from

(ex. Select 'Yes' to enable and provide new values to 'Dialout context' and 'Callback
context' fields)
(Option buttons)
• Volume Gain:
Set the gain you want to apply to VoiceMail recordings.

(ex. Enter 2 to amplify the recordings).
([0-9])
• Dialout context:
Context to dial out from

Example:
Set this field to
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'fromvm'
for example
([a-z][0-9])
• Callback context:
Context to all call back

Example:
Set this field to
'tomv'
If not listed, calling the sender back will not be permitted
([a-z][0-9])
Edit Timezone
Edit Timezone
• Timezone:
Timezone name
(ex. Bosnia and Herzegovina)
(Display)
• Edit
Edits the timezone configuration

(ex. Click to edit the timezone configuration)
(Button)
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• Delete
Deletes a timezone from the system

(ex. Click to delete a timezone from the system)
(Button)
Add/Edit Timezone
Add/Edit Timezone
• Name:
Unique timezone name

(ex. Name provided here will be visible when setting correct voicemail timezone. Type
'Zenica' here for example)
([a-z][0-9])
• Timezone:
Set the correct timezone

(ex. If you have set 'Name'='Zenica' (a town in Bosnia) select the closes timezone to
Zenica here (e.g. 'Europe/Sarajevo'))
(Select box)
• Time format:
Set the appropriate time format

Example:
Depending on selected 'Timezone' you may choose between the following options:
• 12 Hour clock
• 12 Hour clock including minute
• 12 Hour clock AM/PM
• 12 Hour clock AM/PM, including minute
• 24 Hour clock
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• 24 Hour clock including minute
• AM/PM 12 hour syntax
• Dutch syntax
• German syntax
• Greek syntax
• Italian syntax
• Norwegian syntax
• Swedish syntax
(Select box)
• Date format:
Set the correct date format

Example:
Depending on selected 'Timezone' you may choose between the following options
• Month/Day/Year
• Day of Week/Month/Day/Year
• Day/Month/Year
• Day of Week/Day/Month/Year
(Select box)
• Custom sound:
This file is played before the voicemail arrival time

(ex. Enter sound file name, without the extension (e.g. 'arlington') here)
([a-z][0-9])
ADSI
ADSI
• ADSI Feature Number
The ADSI feature descriptor number to download to

(ex. 0000000F)
([A-F][0-9])
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• ADSI Security code
The ADSI security lock code

(ex. 9BDBF7AC)
([A-F][0-9])
• ADSI Application version
The ADSI voicemail application version number

(ex. 1)
([0-9])
Mailbox Option
Calls are diverted to Voicemail when a user is unavailable or has the phone powered off
or when a call is transferred to a voicemail by the user. The phone alerts the user to
indicate the receipt of a message.
Once the user is transferred to the party's voice box the 'Please leave a detail message
1. To leave a voice message that is ended by pressing # key or by hanging up

If 0 is dialed 'Press 1 to accept this recording, otherwise please continue to hold'

message will be heard. User has two options:
1. Press 1 to save your message and dial the operator. 'Please hold while i try that
please hold while i try that extension' message is played.
• 1 Read voicemail messages

• 2 Change folders
o 0 Mailbox options
o 1 Record your unavailable message
o 2 Record your busy message
o 3 Record your name
o 4 Record your temporary message (new in Asterisk v1.2)
o 5 Change your password
o * Return to the main menu
• 3 Advanced options (with option to reply; introduced in Asterisk CVS Head
April 28, 2004 with 'enhanced voicemail')
o 1 Reply
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o 2 Call back(1)
o 3 Envelope
o 4 Outgoing call(1)
• 4 Play previous message
• 5 Repeat current message
• 6 Play next message
• 7 Delete current message
• 8 Forward message to another mailbox
• 9 Save message in a folder
• * Help; during msg playback: Rewind
• # Exit; during msg playback: Skip forward
• ** Help
• *# Exit
After recording a message (incoming message, busy/unavailable greeting, or name)
• 1 Accept
• 2 Review
• 3 Re-record
• 0 Reach operator(1) (not available when recording greetings/name)
Configuration Files
System configuration files are accessible through this section. Only trained users should
modify configuration files.
Configuration Files
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First select a configuration file from the right hand side navigation (e.g. FEATURES).
The file will open in big text box from where it can be modified. Once the file is
changed, click on the 'Save' button.
If there is a message saying 'Configuration files not updated' visible, click on 'Import
current file' or 'Import All Files' button to update the files. This error usually happens if
configuration file is changed through system console (shell).
Tip
Once any kind of file update is done be sure to Reload PBXware MT to apply the
changes.
g729
g729
• Codec optimized for:
Select the server architecture.

(ex. Select the type of server processor)
(Select box)
• Register Utility:
Path to codec's register utility, required for codec install

(ex. Please note that the default path leads to 32-bit software architecture. For 64-bit
please visit http://ftp.digium.com/pub, navigate to desired version and paste the new
destination into this field).
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([a-z] [0-9] [:/_.-])
• Licence Key:
Codec license key:

(ex. Provide the codec license key here).
([a-z] [0-9] [_.-])
• Install:
Install button
(ex. After all necessary details are provided with the correct data(valid urls), click this
button to install the g729 codec).
(Command button)
About
About section displays systems edition, version, release date and licensing information.
About
• PBXware MT:
This line identifies the PBXware MT version, release date (Revision) and Asterisk
running
(ex. Edition: Business, Version: 3.0, Running: 1.4.24-gc-86b7f08)
(Display)
• Package:
Package information displays a number of system Extensions, Trunks, Conferences ...

(ex. Servers: 1, Extensions 768...)
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(Display)
• Enhanced Services:
This line identifies system Enhanced Services

(ex. Last Caller, Group Hunt, Call Forwarding...)
(Display)
• License Details:
This line displays license details. NOTE: If 'Branding' is enabled, only license number is
visible
(ex. License No: 7E5CF50C)
(Display)
Settings: On slave tenants

Settings on a slave are used to set current slaves variables. These settings are applied
only on the current slave tenant.
Settings: On slave tenants
Default Trunks
Slaves use trunks, assigned to them on the master tenant, to place calls to various
destinations. In order to allow an organization to control its voice communications
budget and to provide for termination backup the default trunks allows setting primary,
secondary and tertiary trunks.
The slave tenant will use the primary trunk as its first choice for every destination
called. If the primary trunk for some reason fails to terminate the call, the secondary
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trunk will be used by the system. If the secondary trunk for some reason fails to
terminate the call, the tertiary trunk will be used by the system.
Defaults Trunks
Select default trunks on the system level.

NOTE: If the dialed number is busy, the system will recognize it and won't skip to other
trunks in order to dial it.
(ex. Service provider name)
(Select box)
• Default Destination:
Default destination where trunks without DID will be transferred

(ex. If there is no DID for an incoming Trunk/Provider, all calls will be transferred to
this system number (e.g. 2000))
([0-9])
Precedence
Settings:
• Default Trunks: All System calls go through trunks defined here

• MiniLCR: Overrides 'Default Trunks' and sets a specific trunk for a destination
Extensions:
• Trunks: Overrides 'Settings: Default Trunks'

• Routes: Overrides 'Settings: MiniLCR'
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UAD
UAD (User Agent Devices) are various IP phones, soft phones, ATA (Analog
Telephone Adaptors) and IAD (Integrated Access Devices) used for system extensions.
PBXware MT supports a wide range of UAD using SIP,IAX, MGCP and ZAPTEL
protocols.
Supported devices are already pre-configured with the most common settings in order to
allow administrators an easy way of adding extensions. However, some PBXware MT
installations have specific requirements hence it is advisable to edit the selected UAD
and set it to the required values. Additionally if an installation needs to use an UAD not
listed, clicking on "Add User Agent" allows the addition of a new UAD.
Requirements
Paging
Paging is a service that supports transmitting of messages to multiple phones over their
loudspeakers
GXP 2000
1. Navigate in your Internet browser to the telephone's IP address

2. Provide the admin password ('admin' by default)
3. Click on the 'Login' button
4. Click on the 'Account *' you wish to edit
5. Scroll down to the 'Auto-Answer' options
6. Select 'Yes'
7. Click on the 'Update' button
8. Click on the 'Reboot' button
Tip
Paging tested with Firmware version 1.0.1.12
Snom 190/320

2. Navigate to your line e.g. 'Line 1'
3. Click 'SIP'
4. Set 'Auto Answer' to 'On'
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6. Navigate to 'Preferences'
7. Set 'Auto Answer Indication' to 'On' for a sound to be played notifying you a call has
been received
8. Set 'Type of Answering' to suit your needs e.g. 'Handsfree'
Tip
Paging tested with Firmware version 5.2b
Polycom 30x/50x/60x
You need the latest version of both the SIP software and bootROM to do it. Auto-
answer could be configured only using provisioning. To prepare configuration files, you
have to complete the following steps: 1. In the 'sip.cfg' file, look for the line with these
variables:
<alertInfo voIpProt.SIP.alertinfo.1.value="Auto Answer"

voIpProt.SIP.alertInfo.1.class="3"...>
Polycom calls up class 3 in sip.cfg or ipmid.cfg file. 2. In 'sip.cfg' my ring class

'AUTO_ANSWER' looks like this:
<ringType se.rt.enabled="1" se.rt.modification.enabled="1">

<DEFAULT se.rt.1.name="Default" se.rt.1.type="ring"
se.rt.1.ringer="2" se.rt.1.callWait="6" se.rt.1.mod="1"/>
<VISUAL_ONLY se.rt.2.name="Visual" se.rt.2.type="visual"/>
<AUTO_ANSWER se.rt.3.name="Auto Answer" se.rt.3.type="answer"/>
<RING_ANSWER se.rt.4.name="Ring Answer" se.rt.4.type="ring-answer"
se.rt.4.timeout="2000" se.rt.4.ringer="2" se.rt.4.callWait="6"
se.rt.4.mod="1"/>
<INTERNAL se.rt.5.name="Internal" se.rt.5.type="ring"
<EXTERNAL se.rt.6.name="External" se.rt.6.type="ring"
<EMERGENCY se.rt.7.name="Emergency" se.rt.7.type="ring"
<CUSTOM_1 se.rt.8.name="Custom 1" se.rt.8.type="ring"
</ringType>
'se.rt.3.type="ANSWER"' sets Polycom phone ring type, in this case an answer, that
means that phone will automatically answer without ringing.
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3. Update the modified files to the provisioning server 4. Reload PBXware MT if used
as a provisioning server 5. Restart your telephone The bootROM on the telephone
performs the provisioning functions of downloading the bootROM, the <Ethernet
address>.cfg file, and the SIP application and uploading log files. The SIP application
performs the provisioning functions of downloading all other configuration files,
uploading and downloading the configuration override file and user directory,
downloading the dictionary and uploading log files.
The protocol which will be used to transfer files from the boot server depends on sev-
eral factors including the phone model and whether the bootROM or SIP application
stage of provisioning is in progress. TFTP and FTP are supported by all SoundPoint and
SoundStation phones. The SoundPoint IP 301, 501, 600 and 601 and SoundStation IP
4000 bootROM also supports HTTP while the SIP application supports HTTP1 and
HTTPS. If an unsupported protocol is specified, this may result in unex-pected
behavior, see the table for details of which protocol the phone will use. The "Specified
Protocol" listed in the table can be selected in the Server Type field or the Server
Address can include a transfer protocol, for example http://usr:pwd@server (see
2.2.1.3.3 Server Menu on page 10). The boot server address can also be obtained via
DHCP. Configuration file names in the <Ethernet address>.cfg file can include a
transfer protocol, for example https://usr:pwd@server/dir/file.cfg. If a user name and
password are specified as part of the server address or file name, they will be used only
if the server supports them.
Tip
A URL should contain forward slashes instead of back slashes and should not contain
spaces. Escape characters are not supported. If a user name and password are not
specified, the Server User and Server Password will be used.
UAD
For downloading the bootROM and application images to the phone, the secure HTTPS
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protocol is not available. To guarantee software integrity, the bootROM will only
download signed bootROM or application images. For HTTPS, widely recog-nized
certificate authorities are trusted by the phone and custom certificates can be added. See
6.1 Trusted Certificate Authority List on page 151. Using HTTPS requires that SNTP be
functional. Provisioning of configuration files is done by the application instead of the
bootROM and this transfer can use a secure protocol.
Cisco 7940-7960
1. Create a new line on a Cisco phone, and put the configuration into sip.conf as you
normally would(go into 'Settings: Call Preferences: Auto Answer (intercom)' and then
make the line you've just created as 'auto-answer'. 2. Here are the contents of
/var/lib/asterisk/agi-bin/callall:
#!/bin/sh
cp /var/lib/asterisk/agi-bin/*conf /var/spool/asterisk/outgoing
3. Make sure to make the script executable. And then for every extension you have as an
auto-answer, have a file like this in /var/lib/asterisk/agi-bin:
Channel: SIP/2006
Context: add-to-conference
WaitTime: 2
Extension: start
Priority: 1
CallerID: Office Pager <5555>
So, for example, if you have three lines that are configured for automatic answering -
SIP/2006, SIP/2007, SIP/2008, you should have three files named 2006-conf, 2007-
conf, 2008-conf in /var/lib/asterisk/agi-bin that get copied into the outgoing call spool
directory every time you call extension 5555.
4. Now, dial 5555 from any phone and you should have one-way paging.
People who use the pager may have to get used to waiting 1-2 seconds before speaking
to allow all the phones to catch up with the audio stream. All of the phones hang up
after 20 seconds, regardless of whether the person originating the page has stopped
talking. Change the AbsoluteTimeout values to increase this interval.
If you want a really confusing loud mess, then change the "dmq" options to "dq" and
you'll get an N-way conversation going with everyone who has a phone. This is not
good.
If you want a really interesting office surveillance tool, change the "dmq" to "dt" and
you'll suddenly be listening to all of the extensions in the office, like some kind of
mega-snoop tool. This is useful for after-hours listening throughout the entire office.
Tip
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Paging tested with Firmware version 6.1
Linksys 941

2. Select 'Admin Login'
3. Select 'Advanced'
4. Navigate to the 'User' tab
5. Set 'Auto Answer Page' to 'Yes'
6. Set 'Send Audio To Speaker' to 'Yes'
7. Click on the 'Submit All Changes' button
Tip
At the time, the paging option is set for all lines and works once a handset is picked up.
Paging tested with Firmware version 4.1.12(a)
Aastra 480i/9133i/9112i

2. Select 'Preferences' under 'Basic Settings' tab
3. Set 'Microphone Mute' to 'No'
4. Set 'Auto-Answer' to 'Yes'
5. Click on the 'Save Settings' button
6. Reboot the phone to apply the new settings
Tip
At the time, the paging option is set for all lines and works once a handset is picked up.
Paging tested with Firmware version 1.3.1.1095
Add/Edit
SIP
Click on 'Add User Agent' to add a device or click on the 'Edit' icon next to one to
change its settings.
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SIP
• Protocol:
Select protocol UAD(User Agent Device) uses

(ex. If UAD/Phone uses SIP protocol, select 'SIP' here)
(Select box)
• Device Name:
Unique device name

(ex. AASTRA 480i)
([a-z][0-9])
(ex. This is a sound heard when dialing digits on touch-tone phones. Each phone has
different 'DTMF Mode'. By default, this field is populated automatically for supported
devices. If adding other UAD/Phone select between 'inband', , 'rfc2833' or 'info'
options)
(Select box)
• Context:
as a collection/group of extensions. Default context used by the PBXware MT is
'default' and must not be used by custom extensions.
(ex. default)
([a-z][0-9])
• Status:
Extension status/presence on the network
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(ex. If this field is set to 'Active', this UAD/Phone will be available in the 'UAD' select
box by default when adding new extensions)
(Select box)
• Internal UAD name for Auto-Provisioning
This is where you define how the system will recognize UAD internally
(ex. aa57i)
(Select box)
• NAT (Network ADdress Translation):
Set the appropriate Extension - PBXware MT NAT relation

Example:
If Extension 1000 is trying to register with the PBXware MT from a remote

here:

• never - Never attempt NAT mode or RFC3581 support
• route - Assume NAT, don't send rport
(Option buttons)
• Canreinvite
No.
(ex. Select Yes if you want Asterisk to send reinvite to the client)
(Option buttons)
• Qualify
Timing interval in milliseconds at which a 'ping' is sent to UAD/Phone or trunk, in order

to find out its status(online/offline).
(ex. Set this option to '2500' to send a ping signal every 2.5 seconds to UAD/Phone or
trunk. Navigate to 'Monitor: Extensions' or 'Monitor: Trunks' and check the 'Status'
field)
([0-9])
• Ringtime:
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UAD/Phone ring time
(ex. Time in seconds UAD/Phone will ring before the call is considered unanswered)
([0-9])
• Incoming Dial Options:
Advanced dial options for all incoming calls

(ex. Please see below for detail list of all available dial options( default: tr ))
([a-z])
• Outgoing Dial Options:
Advanced dial options for all outgoing calls

(ex. Please see below for detail list of all available dial options( default: empty ))
([a-z])
• Incoming Limit:
Maximum number of incoming calls

(ex. 2)
([0-9])
• Outgoing Limit:
Maximum number of outgoing calls

(ex. 2)
([0-9])
• Disallow:

(Read only)
• Allow:

(Check box)
• Auto-Framing (RTP Packetization):
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If autoframing is turned on, system will choose packetization level based on remote
ends preferences.
(ex. Yes)
(Option buttons)

(ex. Connect UAD/Phone to PBXware MT without any hassle by providing UAD/Phone
MAC address( and optionally adding Static UAD/Phone IP address and network
details))
(Option Buttons)
• DHCP ( Dynamic Hosts Configuration Protocol ):
Set whether UAD/Phone is on DHCP or Static IP address

fields bellow).
(Option buttons)
• Presence
Presence enables UAD to subscribe to hints which are used for BLF.
(ex. Yes, No, N/A)
(Options buttons)
• User Agent Auto Provisioning Template:
This option is used for providing additional config parameters for SIP, IAX and MGCP
([a-z][0-9])
IAX
Click 'Add User Agent' to add a device or click 'Edit' icon next to one, to change its
settings.
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IAX
• Protocol:
Select protocol UAD (User Agent Device) uses

(ex. If UAD/Phone uses IAX protocol, select 'IAX' here)
(Select box)
• Device Name:
Unique device name

(ex. AASTRA 480i)
([a-z][0-9])
(ex. This is a sound heard when dialing digits on touch-tone phones. Each phone has
different 'DTMF Mode'. By default, this field is populated automatically for supported
devices. If adding other UAD/Phone select between 'inband', , 'rfc2833' or 'info'
options)
(Select box)
• Context:
as a collection/group of extensions. Default context used by the PBXware MT is
'default' and must not be used by custom extensions.
(ex. default)
([a-z][0-9])
• Status:
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(Select box)
• NAT (Network Address Translation):
Set the appropriate Extension - PBXware MT NAT relation

Example:
If Extension 1000 is trying to register with the PBXware MT from a remote

here:

• never - Never attempt NAT mode or RFC3581 support
• route - Assume NAT, don't send report
(Option buttons)
• Canreinvite
No.
(ex. Select Yes if you want Asterisk to send reinvite to the client)
(Option button)
• Qualify
Timing interval in milliseconds at which a 'ping' is sent to UAD/Phone or trunk, in order

to find out its status(online/offline).
(ex. Set this option to '2500' to send a ping signal every 2.5 seconds to UAD/Phone or
trunk. Navigate to 'Monitor: Extensions' or 'Monitor: Trunks' and check the 'Status'
field)
([0-9])
• Ringtime:
UAD/Phone ring time

(ex. Time in seconds UAD/Phone will ring before the call is considered unanswered)
([0-9])
• Incoming Dial Options:
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Advanced dial options for all incoming calls
(ex. Please see below for detail list of all available dial options( default: tr ))
([a-z])
• Outgoing Dial Options:
Advanced dial options for all outgoing calls

(ex. Please see below for detail list of all available dial options( default: empty ))
([a-z])
• Incoming Limit:
Maximum number of incoming calls

(ex. 2)
([0-9])
• Outgoing Limit:
Maximum number of outgoing calls

(ex. 2)
([0-9])
• Notransfer:
Disable native IAX transfer

(Option buttons)
• Send ANI:
Should ANI ("super" Caller ID) be sent over this Provider

(Option buttons)
• Trunk:
Use IAX2 trunking with this host

(Option buttons)
• Auth Method:
Authentication method required by provider

(ex. md5)
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([a-z][0-9])
• Encryption:
Should encryption be used when authenticating with the peer

([a-z][0-9])
• Disallow:

(Read only)
• Allow:

(Check box)

(ex. Connect UAD/Phone to PBXware MT without any hassle by providing UAD/Phone
MAC address( and optionally adding Static UAD/Phone IP address and network
details))
(Option Buttons)
• DHCP ( Dynamic Hosts Configuration Protocol ):
Set whether UAD/Phone is on DHCP or Static IP address

fields bellow).
(Option buttons)
• Presence
Presence enables UAD to subscribe to hints which are used for BLF.
(ex. Yes, No, N/A)
(Options buttons)
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This option is used for providing additional config parameters for SIP, IAX and MGCP
([a-z][0-9])
ZAPTEL
ZAPTEL
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Zapata General
Zapata General
• Device Name:
Unique device name

(AASTRA 480i)
([a-z][0-9])
• Channels
Which card channels are used

(ex. 1,4/1-4)
([0-9][,-])
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• Language:
Default language
(ex. us)
(Select box)
• Status:

(Select box)
• Signaling:
Signaling method
Example:
Default
• FXS Loopstart
• FXS Groundstart
• FXS Kewlstart
• FXO Loopstart
• FXO Groundstart
• FXO Kewlstart
• PRI CPE side
• PRI Network side
• BRI CPE side
• BRI Network side
• BRI CPE PTMP
• BRI Network PTMP
(Select box)
• Music On Hold:
Select which class of music to use for music on hold. If not specified, the 'default' will
be used
(ex. default)
(Select box)
• Mailbox:
Define a voicemail context

(ex. 1234, 1234@context)
376
([a-z][0-9])
PRI
PRI
• Switchtype:
Set switch type

Example:
• National ISDN 2
• Nortel DMS100
• AT&T 4ESS
• Lucent 5ESS
• EuroISDN
• Old National ISDN 1
(Select box)
• PRI Dial Plan:
Set the dial plan used by some switches

Example:
• Unknown
• Private ISDN
• Local ISDN
• National ISDN
(Select box)
• PRI Local Dial Plan:
Set the numbering dial plan for destinations called locally

Example:
• Unknown
• Private ISDN
377
• Local ISDN
• National ISDN
(Select box)
• PRI Trust CID:
Trust the provided caller id information

(ex. Yes, No, N/A)
(Option buttons)
• PRI Indication:
How to report 'busy' and 'congestion' on a PRI

Example:
• outofband - Signal Busy/Congestion out of band with RELEASE/DISCONNECT

• inband - Signal Busy/Congestion using in-band tones
(Select box)
• Network Specific Facility:
If required by the switch, select the network specific facility

Example:
• none
• sdn
• megacom
• accunet
(Select box)
Caller ID
378
Caller ID
• Outbound Caller ID:
Caller ID set for all outbound calls where Caller ID is not set or supported by a device
([0-9])
• Caller ID:
CallerID can be set to 'asreceived' or a specific number if you want to override it.
NOTE: Caller ID can only be transmitted to the public phone network with supported
hardware, such as a PRI. It is not possible to set external caller ID on analog lines
(ex. 'asreceived', 555648788)
([a-z][0-9])
• Use Caller ID:
Whether or not to use caller ID

(ex. Yes, No, N/A)
(Options buttons)
• Hide Caller ID:
Whether or not to hide outgoing caller ID

(ex. Yes, No, N/A)
(Options buttons)
• Restrict CID:
switch is sending
(ex. Yes, No, N/A)
(Options buttons)
• CID Signaling:
Set the type of caller ID signaling

Example:
• bell - US
• v23 - UK
• dtmf - Denmark, Sweden and Netherlands
379
(Select box)
• CID Start:
What signals the start of caller ID

Example:
• ring = a ring signals the start

• polarity = polarity reversal signals the start
(Select box)
• Call Waiting CID:
Whether or not to enable call waiting on FXO lines

(ex. Yes, No, N/A)
(Options buttons)
• Send CallerID After:
Some countries, like the UK, have different ring tones (ring-ring), which means the
caller ID needs to be set later on, and not just after the first ring, as per the default.
(ex. Yes)
(Select box)
Echo Canceller
Echo Canceller
• Echo Cancel:
Enable echo cancellation

(ex. Yes, No, N/A)
(Option Button)
• Echo Training:
Mute the channel briefly, for 400ms, at the beginning of conversation, cancelling the
echo. (Use this only if 'Echo Cancel' doesn't work as expected)
380
(ex. Yes, No, N/A)
(Option buttons)
• Echo Cancel When Bridged:
Enable echo cancellation when bridged. Generally not necessary, and in fact
undesirable, to echo cancel when the circuit path is entirely TDM
(ex. Yes, No, N/A)
(Option buttons)
Call Features
Call Features
• Call Waiting:
Whether or not to enable call waiting on FXO lines

(ex. Yes, No, N/A)
(Option buttons)
• Three Way Calling:
Support three-way calling. If enabled, the call can be put on hold and one is able to
make another call
(ex. Yes, No, N/A)
(Option buttons)
• Transfer:
Support call transfer and also enables call parking (overrides the 'canpark' parameter).
Requires 'Three Way Calling' = 'Yes'.
(ex. Yes, No, N/A)
(Option buttons)
381
• Can Call Forward:
Support call forwarding

(ex. Yes, No, N/A)
(Option buttons)
Call Return:
Whether or not to support Call Return '*69'. Dials last caller extension number
(ex. Yes, No, N/A)
(Option buttons)
• Overlap Dial:
Enable overlap dialing mode (sends overlap digits)

(ex. Yes, No, N/A)
(Option buttons)
• PulseDial:
Use pulse dial instead of DTMF. Used by FXO (FXS signaling) devices
(ex. Yes, No, N/A)
(Option buttons)
Call Indications
Call Indications
• Distinctive Ring Detection:
Whether or not to do distinctive ring detection on FXO lines

(ex. Yes, No, N/A)
(Option buttons)
• Busy Detect:
382
Enable listening for the beep-beep busy pattern
(ex. Yes, No, N/A)
(Option buttons)
• Busy Count:
How many busy tones to wait before hanging up. Bigger settings lower the probability
of random hangups. 'Busy Detect' has to be enabled
Example:
• 4
• 6
• 8
(Select box)
• Call Progress:
Easily detect false hangups

(ex. Yes, No, N/A)
(Option buttons)
• Immediate:
Should channel be answered immediately or the simple switch should provide

dialtone, read digits, etc.
(ex. Yes, No, N/A)
(Option buttons)
Call Groups
Call Groups
• Call Group:
Set to which the Call Group extension belongs.

(ex. Similar to 'Context' grouping, only this option sets to which call group extension
belongs(Allowed range 0-63))
383
([0-9] [,-])
• Pickup Group:
Set group's extension as allowed to pick up.

(ex. Similar to 'Context' grouping, only this option sets the Call Groups extension as
allowed to pick up by dialing '*8').
([0-9] [,-])
RX/TX
RX/TX
• RX Wink:
Set timing parameters

Example:
• Pre-wink (50ms)
• Pre-flash (50ms)
• Wink (150ms)
• Receiver flashtime (250ms)
• Receiver wink (300ms)
• Debounce timing (600ms)
(Select box)
• RX Gain:
Receive signal decibel

(ex. 2)
([0-9])
• TX Gain:
Transmit signal decibel

(ex. 2)
([0-9])
384
• ADSI (Analog Display Services Interface):
Enable remote control of screen phone with softkeys. (Only if you have ADSI
compatible CPE equipment)
(ex. Yes, No, N/A)
(Option buttons)
• Jitter Buffers:
Configure jitter buffers. Each one is 20ms long

(ex. 4)
([0-9])
• Relax DTMF:
If you are having trouble with DTMF detection, you can relax the DTMF detection
parameters
(ex. Yes, No, N/A)
(Option buttons)
• Fax Detect:
Enable fax detection

Example:
• both
• incoming
• outgoing
• no
(Select box)
385
Span
Span
• Span number:
Number of the span

(ex. 1)
([0-9])
• Span timing:
How to synchronize the timing devices

Example:
• 0 - do not use this span as sync source

• 1 - use as primary sync source
• 2 - set as secondary and so forth
([a-z])
• Line build out:
Example:
• 0 db (CSU) / 0-133 feet (DSX-1)

• 133-266 feet (DSX-1)
• 266-399 feet (DSX-1)
• 399-533 feet (DSX-1)
• 533-655 feet (DSX-1)
• -7.5db (CSU)
• -15db (CSU)
• -22.5db (CSU)
(Select box)
• Framing:
How to communicate with the hardware at the other end of the line
Example:
386
• For T1: Framing is one of d4 or esf.
• For E1: Framing in one of cas or ccs.
(Select box)
• Coding:
How to encode the communication with the other end of line hardware.
Example:
• For T1: coding is one of ami or b8zs

• For E1: coding is one of ami or hdb3 (E1 may also need crc)
(Select box)
• Yellow:
Whether yellow alarm is transmitted when no channels are open.

(ex. Yes, NO, N/A)
(Option buttons)
Dynamic Span
Dynamic Span
• Dynamic span driver:
The name of the driver (e.g. eth)
• Dynamic span address:
Driver specific address (like a MAC for eth).
• Dynamic span channels:
Number of channels
• Dynamic span timing:
387
Sets timing priority, like for a normal span. Use "0" in order not to use this as a timing
source, or prioritize them as primary, secondary, etc.
FXO Channels
FXO Channels
• FXO Loopstart:
Channel(s) are signaled using FXO Loopstart protocol
• FXO Groundstart:
Channel(s) are signaled using FXO Groundstart protocol
• FXO Kewlstart:
Channel(s) are signaled using FXO Kewlstart protocol
The values for the above fields are set as follows:
• 1 - for one card

• 1-2 - for two cards
• 1-3 - for three cards etc.
• 2-3 (If your card has modules in this order FXS, FXO, FXO, FXS)
FXS Channels
FXS Channels
• FXS Loopstart:
Channel(s) are signaled using FXS Loopstart protocol
388
• FXS Groundstart:
Channel(s) are signaled using FXS Groundstart protocol
• FXS Kewlstart:
Channel(s) are signaled using FXS Kewlstart protocol
Values for the above fields are set as follows:
• 1 - for one card

• 1-2 - for two cards
• 1-3 - for three cards etc.
• 2-3 (If your card has modules in this order FXS, FXO, FXO, FXS)
PRI Channels
PRI Channels
• D-Channel(s):
For example, every ISDN BRI card has 1 D- (control) channel

(ex. 1)
([0-9])
• B-Channels(s):
For example, every ISDN BRI card has 2 B- (data) channels

(ex. 2)
([0-9])
Other Zaptel Channels
Other Zaptel Channels
389
• Unused:
([0-9])
• Clear:
([0-9])
WOOMERA
WOOMERA
• Protocol:
Select protocol UAD(User Agent Device) uses

(ex. If UAD/Phone uses MGCP protocol, select 'MGCP' here)
(Select box)
• Device Name:
Unique device name

(ex. AASTRA 480i)
([a-z][0-9])
• Status:

(Select box)
390
This option is used for providing additional config parameters for WOOMERA
configuration files. The values provided here will be written into these configuration
files.
([a-z][0-9])
Access Codes
Access codes provide tenant user with access to essential system or enhanced services
Access Codes
• Voicemail:
Voice inbox access code. This number is dialed to access the extension voice inbox
(extension PIN required)
(ex. From extension 1000, dial '*123' to access the extension 1000 voice inbox. When
asked for a PIN, provide the PIN set for this extension)
([0-9])
• General Voicemail:
Access code for general voice mailbox. This number is used for checking your voice
inbox from any system Extension
(ex. Dial '*124'. Enter your Extension number and PIN when asked for it)
([0-9])
• Voicemail Transfer:
Access code for transferring active calls to any system voice box
391
(ex. During active conversation dial '*125 + $EXTENSION' to transfer calling party to
system $EXTENSION number voice box)
([0-9])
• Agent Pause:
Access code for pausing an agent.

(ex. If agent dials Agent Pause access code, his extension is not going to receive any
calls until he dials the same code to unpause itself)
([0-9])
• Agent Not Ready:
Access code for setting agent to Not Ready.

(ex. If agent dials Agent Not Ready code, his extension is not going to receive any calls,
while listening MOH, until he dials the same code again).
([0-9])
• Last Caller:
Access code for dialing the last Extension that called you
(ex. If Extension '1000' was the last extension that called you, dial '*149' from your
Extension. Message 'The number to call your line was $EXTENSION. To call this number
press 1' will be played. Press '1' dial this destination).
([0-9])
• Monitoring:
Access code for monitoring active calls

(ex. If Extensions '1000' and '1001' are in conversation, dial '*199 + 1000' to listen the
ongoing conversation)
([0-9])
• Monitor Conferences:
This is the access code for Conference monitoring, and its usage is described in
Enhanced Services of the extensions
(ex. *500+CONF_NUM)
([0-9] *)
• Speakerphone Page:
392
Access code for transmitting a message to multiple phones via their loudspeakers
(ex. Dial '*399' to speakerphone all Extensions defined under Enhanced Services)
([0-9])
• Single Speakerphone Page:
Access code for transmitting a message to phone loudspeaker

(ex. Dial '*400 + $EXTENSION' to transmit a message to provided extension (phone)
loudspeaker)
([0-9])
• Speed Dial:
Speed Dial enables faster dialing by entering access code and dual number speed
dialing code. Speed Dialing Codes can be entered by clicking Settings: Servers and then
clicking on Speed Dial button
(ex. Dial '*130 + XX' where XX represents speed dialing code associated to some
extension)
([0-9])
• Other Networks:
Access code for accessing other PBXware MT networks

Example:
Dial '*188 + $NETWORK + $EXTENSION' to dial number $EXTENSION or $NETWORK

(e.g. '*188 8 1000').
NOTE: 'Trunk' and 'Other Network' must already be set
([0-9])
• Listen to CDR recordings:
Access code used when you want to listen last 9 Call Recordings from CDRs
(ex. Dial '*170' access code followed by a number between 1 and 9, where 1 is most
recent recorded conversation (excluding access code calls)).
([0-9])
• Enable Call Forwarding:
Access code for enabling Call Forwarding Enhanced Service

(ex. Dial '*71 + $EXTENSION' to forward all calls to $EXTENSION number. This number
can be local Extension or Proper/Mobile number (e.g. '*71 1001' or '*71 55510205'))
393
([0-9])
• Disable Call Forwarding:
Access code for disabling Call Forwarding Enhanced Service

(ex. Dial '*72' to disable this service (Extension number not required))
([0-9])
• Block CallerID:
Access code to block other users from seeing your CallerID

(ex. Dial '*67' to block displaying your CallerID)
([0-9])
• Block CallerID once:
Access code for blocking only first next call from displaying CallerID
(ex. Dial '*81' to block only next call from seeing your CallerID)
([0-9])
• Unblock CallerID:
Access code to unblock your CallerID after it has been blocked

(ex. Dial '*68' to unblock your CallerID)
([0-9])
• Enhanced Call Park:
Access code for parking calls and sending them to preset Announce Extension which
can be set in Settings->Servers
(ex. During active conversation dial '#800'. The call will be parked and Announce
Extension will ring for Timeout seconds. After that period call will be directed to
Timeout Extension)
([0-9])
• Music On Hold:
Access code for playing Music On Hold sound files

(ex. Dial '*388' to play 'default' Music On Hold class sound files)
([0-9])
• Echo Audio Read:
394
Access code for echo audio test
(ex. Dial '*398' and talk. Everything you say is returned back so the server response
time can be checked)
([0-9])
• Record Greeting:
Record greeting or any other sound file that can be used in PBXware MT
(ex. Dial '*301' and after the beep say the message that you want to record)
([0-9])
• Agent Greeting:
Access code for recording agents greeting message which is played every time agent
receives a call.
(ex. Agent dials this code, leaves a message which is then played to all his callers).
([0-9])
• Queue Interrupt Message:
Record an interrupt message to be played to callers waiting in queue

(ex. When a user on the system dials this access code, system will ask him to enter
agent number, PIN, and a queue to which he wants to play the message. When that
info is entered user will record a message after which he will press #. System will ask if
you want to review your message, record it again or confirm recorded message. When
you confirm message it will be played to all callers waiting in given queue).
([0-9])
• Open Operation Times:
User will dial '*401' to open systems operation times

(ex. If operation times is open, but not explicitly closed, it is automatically closed at 12
AM)
([0-9])
• Close Operation Times:
User will dial '*402' to close systems operation times

(ex. If user doesn't close operation times, it will be closed automatically at 12 AM)
([0-9])
395
Numbering Defaults
Numbering defaults sets the way PBXware MT will assign network numbers to
Extensions, Conferences etc...
Numbering Defaults
Fetch least unallocated number (default):
This option takes the lowest number on the system that is not used by the system and
assigns it to a new Extension you're trying to create for example.
Fetch next unallocated number:
If the last allocated number was assigned to extension 2010, the next IVR that you're
trying to add for example will be given number 2011. The system will not try to give
you an unallocated number 1022 for example
Fetch random:
Use this option in case that you want to assign numbers in non-sequential order. If you
create a new Queue for example, PBXware MT will assign it a number 2350, for
example. And if you try to create a new Extension right after that, PBXware MT might
assign it a number 9838, for example.
About
About section displays the system's edition, version, release date and licensing
information.
396
About
• PBXware MT:
This line identifies the PBXware MT version, release date (Revision) and Asterisk
running
(ex. Edition: Business, Version: 2.0.0, Release: 2007-06-27 (#1, $Revision: 2135 $),
Running: 1.2.13-b20070521)
(Display)
• Package:
Package information displays a number of system Extensions, Trunks, Conferences,

etc.
(ex. Servers: 1, Extensions 768...)
(Display)
• Enhanced Services:
This line identifies the system's Enhanced Services

(ex. Last Caller, Group Hunt, Call Forwarding...)
(Display)
• License Details:
This line displays license details.

NOTE: If 'Branding' is enabled, only the license number is visible
(ex. License No: 7E5CF50C)
(Display)
397
MT 3.8.5 Site Settings
Contents
• 1 Site Settings
o 1.1 Site Users
o 1.2 Groups
o 1.3 Sessions
o 1.4 Date/Time Settings'
o 1.5 Language
o 1.6 Updates
o 1.7 Licensing
o 1.8 API Key
o 1.9 About
• 2 SM Settings
o 2.1 Network Administrators
o 2.2 IP Restrictions
• 3 Self-Care
o 3.1 Login
o 3.2 Administration Interface
• 4 Help
• 5 Logout
398
Site Settings
Site settings set options such as site users, user groups, backup, updates, and upgrades
options
Sie Settings
Site Users
Site users are allowed to login to the system interface in order to perform a specific
function according to granted permissions. Each user belongs to a user group. Each
group's permissions are pre-set in order to allow unified access and permission control.
User can have access to any application or part of that application depending on
permissions granted. It is highly recommended to add/edit groups before adding new
users.
Add/Edit Users:
399
Ste Users
Add/Edit User
User fields are standard fields required to be entered in order for the system to allow
user access to various applications. In addition, the user's status can be changed to
"suspended" by ticking the check box and pressing "save"
Groups
Groups allow for a unified permission system, enabling users access to various
applications or part of the applications. The system is preset with "common" groups:
Sales, Support, Accounts and Billing, Management.
400
Groups
Each site can edit existing or add new groups as per their requirements by clicking on
the appropriate action buttons. During add/edit permissions and group name is available
for edit.
Group Management
401
Show Advanced
If this box is checked, all users belonging to this group will be able to view and edit all
advanced options fields within the system.
Server
If this box is checked, all users belonging to this group will have administration access
to the server. This allows network administrator to delegate the administration across
the organization.
Reload/Start/Restart/Stop PBXware MT buttons are options that control PBXware MT

actions. If enabled, these controls will be shown on the right side of browser interface.
Groups/Server
Tip
In PBXware MT 3.0 Restart button will not restart all PBXware MT services, only
asterisk service.
If you want to restart PBXware MT service, go to Setup Wizard and click on Services
button which will allow this.
402
Sessions
After a user logs into the system, all data sent between the user and the system can be in
plain text or encrypted by industry standard SSL (secure layer socket). It is worth
mentioning that SSL uses much more resources hence slower browser responses.
Sessions
• Session Type
Session type
Example:
Available options:
• HTTPS only - Complete data flow between system and GUI user is secured
using SSL
• HTTP with HTTPS authentication - Authentication is done using SSL while
normal data flow between GUI user and system is plain text
• HTTP only (including authentication) - Complete data flow between system
and GUI user is plain text
(Option buttons)
Date/Time Settings'
Date/Time Settings
403
• Date Format:
Set the proper date format shown throughout the system's interface
(ex. 04 Oct 2006)
(Select box)
• Time Format:
Set the proper time format shown throughout the system's interface
(ex. Select between 12/24 hour format)
(Select box)
Language
Here you can choose which language you want your Site Manager to display.
Language
Updates
This section allows a shortcut to the licensing screen. It is useful if a license upgrade
needs to be performed.
404
Updates
• Username:
Username used for update.

(ex. This is always pbxware, do not change this).
([a-z][0-9])
• Password:
Password used by user for update.

(ex. This is always update, do not change this).
([a-z][0-9])
• Version:
If you are using older version of PBXware MT, this is the option where you will upgrade
it to newer one
Licensing
Read Getting Started->Setup Wizard->Licensing chapter.
API Key
Here you can generate a random key used to interface with PBXware MT using
PBXware MT API. Additional info on API can be obtained by clicking on
Documentation button.
405
API Key
About
About shows the Site Manager's current release date present on the system.
About
SM Settings
SM Settings gives you the ability to add/edit system administrators and control access to
GUI with IP restrictions.
406
SM Settings
Network Administrators
In this menu you can add a new GUI administrator or edit existing one.
Network Administrators
• Name:
Name of the new admin user

(ex. John)
([a-z])
• Email:
Email address used for admin login

([a-z] [0-9] [@_.-])
• Password:
407
Password used for logging in
(ex. fjhoe5!4fh8o%e54fg_vh8)
([0-9][a-z])
• Verify Password:
Verify given password

(ex. fjhoe5!4fh8o%e54fg_vh8)
([0-9][a-z])
IP Restrictions
The IP restrictions menu is used to set whether some IP ranges have access or are
banned, which depends on selection of the Blacklist/Whitelist in the Settings menu.
Policies
Add/Edit IP restrictions policies which dictate which IP ranges can/cannot access GUI
of PBXware MT.
Policies
When adding/editing, you will provide name and IP range for the current policy.
408
Policies
• Policy Name:
Name that describes this policy

(ex. admin)
([a-z][0-9])
• Policy Range:
IP range on which admin will or will not have access to GUI

(ex. 10.1.0.0/24)
([0-9])
Settings
Select whether set policies will be whitelisted or blacklisted which means they will be
either allowed to access GUI or they will not.
Settings
409
Self-Care
Self-Care is an extension of the administration interface used by the extension owner
Self-Care
Login
In order to login to Self-Care, point your browser to: http://$IPADDRESS/ (For
Example: http://192.168.1.1/)
Login
• Email:
410
Email address assigned to the extension
(ex. Provided email address is used as a username for logging into Self-Care (e.g.
[email protected]))
([a-z] [0-9] [@_.-])
• Password/PIN:
PIN assigned to extension

(ex. This field accepts extension PIN (e.g. 1981))
([0-9])
Extension Control
The user can monitor multiple extensions through the Self-Care interface. To administer
a different extension, select its network number from the 'Select an extension' select
box.
Extension Control
You will be asked to authenticate by providing the extension PIN number. If the correct
extension PIN is provided, the user will administer the selected extension.
Extension Control
411
My Details
User can manage his email account and PIN associated with his extension.
Tip
If the user has a voicemail account only(no system extension), this feature will be
disabled.
Extension Control
• E-mail:
E-mail address associated with the extension. This address is used for various system
notifications and for the user logging into Self-Care
(ex. To login to Self-Care, type this email address into the 'E-mail' field)
([a-z] [0-9] [@._-])
• PIN (Personal Identification Number):
Four digit password used for accessing voicemail and other additional PBXware MT
services as well as logging into Self Care
(ex. To login to Self-Care, type this number into the 'PIN' field)
([0-9])
Directory
If the directory is enabled in the Settings->Servers menu in the administration GUI, you
will see it in the OSC of any extension. The directory will show any extension's name
and number if that extension has the 'Show in Directory' option set to Yes.
412
Directory
• Search
If you have many extensions in a Directory you can search them by Name, E-mail or
Extension number
([a-z][0-9])
• Name
Whether to allow searching by extension names

(Check box)
• E-mail
Whether to allow searching by e-mails of extensions

(Check box)
• Extension
Whether to allow searching by extension number

(Check box)
Tip
To see directory from Polycom phones, one must first create symbolic link of
'/home/sitemanager/admin/apps/pbxware/dir' to '/admin/public_html/'. Later restart
polycom, go to setup and enter server type: Trivial FTP and Server adress, under Server
Menu. Wait for phone to boot, press Applications button to see directory. To call
number from directory just go to wanted extension with up and down arrows and press
confirm.
Voicemail
User can manage voice messages left on his extension from this location.
413
Voicemail
• Msg:
Voicemail message identification number

(ex. 0000)
(Display)
• Caller:
Identifies the user who has left the message by his name and extension number
(ex. "BobReilly" <5000>)
(Display)
• Date:
Time/Date a voicemail has been received in inbox

(ex. 13 Apr 2006 15:12)
(Display)
• Duration:
Time duration of voice message

(ex. 00:18)
(Display)
• Type:
Voicemail file type and size

(ex. wav49 (9.07k))
(Display)
Tip
• wav49 = 91.0kb
• wav = 863.0kb
414
• gsm = 91.0kb
• wav49 = 0.38kb
• wav = 3.0kby
• gsm = 0.32kb
Actions
Actions
• Open:
Displays the content of a voice inbox folder

(ex. Select a destination folder in a Select box and click this button to display its
contents)
(Command Button)
• Move:
Moves the content into a different voice mailbox location/directory

(ex. Select the box next to a voice message, set the destination folder in the Select box
and click this button to move the voice message to a new destination/folder)
(Command Button)
• Forward:
Forward the voice message to another network extension inbox

(ex. Select the box next to a voice message and click this button. When prompted for
the extension, type '1005' for example, and the selected voice message will be
transferred to the voice inbox of network extension 1005)
(Command Button)
• Play:
Downloads/Plays the voice message

(ex. Select the box next to a voice message and click this button to download the
message on the Desktop or to play it in your favorite media player (depending on the
option selected in the popup window))
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(Command Button)
• Delete:
Deletes the voce message from the inbox

(ex. Select the box next to a voice message and click this button to permanently delete
the voice message from the inbox)
(Command Button)
Voicemail
Once the user is transferred to the party's voice box, the 'Please leave a detailed message
1. To leave a voice message that is ended by pressing # key or by hanging up

If 0 is dialed 'Press 1 to accept this recording, otherwise please continue to hold'

message will be heard.
User has two options:
1. Press 1 to save your message and dial the operator. 'Please hold while I try that
please hold while I try that extension' message played.
Standard voicemail options with all voicemail settings set to 'Yes':
• 1 Read voicemail messages

• 2 Change folders
o 0 Mailbox options
o 1 Record your unavailable message
o 2 Record your busy message
o 3 Record your name
o 4 Record your temporary message (new in Asterisk v1.2)
o 5 Change your password
o * Return to the main menu
• 3 Advanced options (with option to reply; introduced in Asterisk CVS Head
April 28, 2004 with 'enhanced voicemail')
o 1 Reply
o 2 Call back(1)
o 3 Envelope:
o 4 Outgoing call(1)
• 4 Play previous message
• 5 Repeat current message
• 6 Play next message
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• 7 Delete current message
• 8 Forward message to another mailbox
• 9 Save message in a folder
• * Help; during msg playback: Rewind
• # Exit; during msg playback: Skip forward
• * * Help
• * # Exit
After recording a message (incoming message, busy/unavailable greeting, or name)
• 1 Accept
• 2 Review
• 3 Re-record
• 0 Reach operator(1) (not available when recording greetings/name)
Enhanced Services
Enhanced Services
Depending on what features are selected by the administrator in the extension's

Enhanced Services, the checked entries will be shown in Online Self Care
Tip
If the user has a voicemail account only, and no system extension, this feature will be
disabled.
CDR
CDR (Call Detail Records) for all placed or received calls on the system. In addition to
normal operation an authorized user is able to perform additional actions such as
extensive search, listen to recorded calls, call any destinations listed and access
advanced features.
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CDR
• From:
Extension number the call was made from

(Display)
• To:
Extension number the call was made to

(Display)
• Date/Time:

(ex. 04 Oct 2006 10:44:10)
(Display)
• Duration:

(ex. 00:12:45)
(Display)
• Billing:

(ex. 00:12:45)
(Display)
• Cost:
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Total cost of the call calculated through a service plan
(ex. 0.71)
(Display)
• Routes:
Number of system routes

(ex. 0.71)
(Display)
• Status:

Example:
following content:
• Answered
• Not Answered
• Busy
• Error
(Display)
service is active
Tip
• wav49 = 84.5kb
• wav = 833.0kb
• gsm = 85.0kb
• wav49 = 84.0kb
• wav = 827.0kb
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• gsm = 84.0kb
Search/Filter
Search/Filter
• Start Date:

(Option button)
• End Date:

(Option button)
• Start Time:
Enter a Search/Filter start time

(ex. 01:20:00)
([0-9] :)
• End Time:
Enter a Search/Filter end time

(ex. 18:20:00)
([0-9] :)
• Status:

Example:
• All
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• Answered
• Not Answered
• Busy
• Error
(Select box)
• Type:
Search calls based on the type of calls

Example:
Click the 'Type' button and select one of the available fields:
• All
• Outgoing
• Incoming
(Select box)
Tip
After making any changes to search filter, be sure to click the search icon
Actions
Call
To establish a call between any PBXware MT extension and a listed extension, you
have to provide only two things: the Caller $EXTENSION number and the
$DESTINATION extension
• Caller
PBXware MT extension that will make a call

Example:
Provide any PBXware MT extension number here, 1001 for example
([0-9])
• Destination:
Destination extension that will be dialed by 'Caller' extension

(ex. To select a destination extension, first check the box next to a CDR record. This
field will display two extensions listed under 'From' and 'Destination' selected record)
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(Select button)
Tip
After setting 'Caller' and 'Destination' extensions click the call icon
Print
new popup window with the printing interface.
Print
Email
Check the box next to a call record and click the 'Email' button. A small popup dialog
will appear. Provide email address here and click on the 'OK' button to send the records.
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Email
Settings
These options mimic the functions of an answering machine but with many additional
features added. Voice messages are saved on central file-system location instead on a
UAD/Phone.
Accessing voice-box:
To access voice-box dial '*123', enter extension PIN and follow the instructions.
Leaving a voice message:
When a user is transferred to an extension's voice-box, the 'Please leave a detail

message after the tone. If you would like to speak to the operator, press 0' message will
be heard. Two options are available:
1. Leave a voice message(ended by pressing '#' key or hanging up)

2. Reach an operator by dialing '0'
If '0' is dialed, the 'Press 1 to accept this recording, otherwise please continue to hold'
message will be heard.
Two options are available:
1. Press '1' to save your message, after which the operator will be dialed. 'Please hold
while I try that extension' message will be heard
2. Continue to hold, which will delete any left messages, after which the operator will be
dialed. 'Message deleted, please hold while I try that extension' message will be heard.
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Settings
• Send E-mail:
Whether or not to send e-mail to the address given in the extensions settings in admin
mode
(ex. Yes, No, N/A)
(Option button)
• Pager e-mail:
Pager e-mail address associated with the voice box.

(ex. When A calls B and leaves a voice message, B will get a pager email notification
about new voice message).
([a-z] [0-9] [@._-])
Greeting message played to users upon entering the voice box.
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(ex. When A gets to B's voice box, the selected 'Greeting message' is played to A
before he is allowed to leave a message).
(Select box)
• Unavailable message:
Upload unavailable message

(ex. Click on the 'Browse' button and select a sound file from local computer to upload
it as a custom unavailable message).
(Button)
• Reset Unavailable message:
Resets the user recorded/uploaded unavailable message.

(ex. Custom unavailable messages can be recorded through UAD/Phone or uploaded
to voice box through Self Care. To revert to default system unavailable message select
'Yes' and save the extension settings).
(Option buttons)
• Busy message:
Upload busy message

(ex. Click on the 'Browse' button and select a sound file from local computer to upload
it as a custom busy message).
(Button)
• Reset Busy message:
Resets the user recorded/uploaded busy message.

(ex. Custom busy messages can be recorded through UAD/Phone or uploaded to voice
box through Self Care. To revert to default system busy message select 'Yes' and save
the extension settings).
(Option buttons)
• Skip Instructions:
Skip the instructions on how to leave a voice message.

(ex. Once user A reaches the dialed voice box, if this option is set to 'Yes', A will hear
the 'Greeting message', and then be transferred directly to the 'beep' sound).
(Option buttons)
• Attach:
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Send the voice message as an attachment to user email.
(Option buttons)
Delete voice message after sending it as an attachment to user email.

be deleted from the voice box after it has been emailed to B)
(Option buttons)
• Say Caller ID:
(ex. If this option is set to 'Yes', when checking voicemail, 'From phone number
(Option buttons)
box.
Example:

(Option buttons)
• Allow Operator:

Example:
following options):
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(Option buttons)
• Operator Extension:

([0-9])
recorded.
(ex. Once voice box is checked for new messages, if this option is set to 'Yes', 'Received
at {$DATE}. From phone number {$NUMBER}' will be played, giving more details about
the message originator).
(Option buttons)
If this option is turned on, callers will not be able to access this extension from IVRs
directory
(ex. Yes, No, N/A)
(Option button)
Number of rings before caller will get voicemail

(ex. 3)
([0-9])
How long to pause in seconds, before asking user for PIN/Password.

(ex. Some UADs/Phones have tendency to garble the beginning of a sound file.
Therefore, user checking the voice box, when asked for password would hear '...sword'
instead of 'Password'. Setting this field to 1-2 seconds will provide long enough gap to
fix this anomaly).
([0-9])
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• Timezone:
Sets the correct date/time stamp.

(ex. By setting the correct time zone, user would always be notified of the exact
date/time voice message was left on their box. Set the correct time zone if user is
located in different time zone then PBXware MT).
(Select box)
Tip
Timezones are taken from '/usr/share/zoneinfo' system directory
Tip
• wav49 = 91.0kb
• wav = 863.0kb
• gsm = 91.0kb
• wav49 = 0.38kb
• wav = 3.0kb
• gsm = 0.32kb
Destinations
If an extension has Destinations set in the Online Self Care option in its Service Plan, it
will be shown in the OSC of the extension.
Help
A click on 'Help' button opens PBXware MT online support files
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Logout
Logs user out of Self-Care.
MT 3.8.5 End Points

This site contains list of UADs supported in PBXware 3.8, also on this page you will
find user manuals for these devices.
Contents
• 1 Supported UAD List
o 1.1 Polycom
o 1.2 Yealink
o 1.3 Cisco
o 1.4 Cisco SPA
o 1.5 Linksys
o 1.6 Sipura
o 1.7 Aastra
o 1.8 Snom
o 1.9 Grandstream
• 2 User Manuals
o 2.1 Polycom
o 2.2 Yealink
o 2.3 Cisco
o 2.4 Linksys
o 2.5 Sipura
o 2.6 Aastra
o 2.7 Snom
o 2.8 Grandstream
o 2.9 xten_networks
o 2.10 Digium
Supported UAD List

Polycom
• Polycom IP 301
• Polycom IP 320
• Polycom IP 321
• Polycom IP 330
• Polycom IP 331
• Polycom IP 335
• Polycom IP 550
• Polycom IP 501
• Polycom IP 550
• Polycom IP 560
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• Polycom IP 601
• Polycom IP 650
• Polycom IP 670
Yealink
• Yealink T18P
• Yealink T20P
• Yealink T22P
• Yealink T26P
• Yealink T28P
• Yealink T32P
• Yealink T38P
• Yealink T42G
• Yealink T60P
• Yealink T65P
• Yealink W52P
Cisco
• Cisco 7940
• Cisco 7941G
• Cisco 7942G
• Cisco 7945G
• Cisco 7960
• Cisco 7961
• Cisco 7962
• Cisco 7965G
• Cisco 7971G
• Cisco 7975G
Cisco SPA
• Cisco SPA301
• Cisco SPA501G
• Cisco SPA502G
• Cisco SPA504G
• Cisco SPA508G
• Cisco SPA509G
• Cisco SPA525G
• Cisco SPA901
Linksys
• Linksys SPA-941
• Linksys SPA-942
• Linksys SPA-962
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Sipura
• Sipura SPA-841
Sipura ATA devices
• Sipura SPA-1000
• Sipura SPA-2000
• Sipura SPA-3000
Aastra
• Aastra 480i
• Aastra 53i
• Aastra 55i
• Aastra 57i
• Aastra 9112i
• Aastra 9133i
Snom
• Snom 190
• Snom 320
• Snom 360
Grandstream
• Grandstream GXP-2000
• Grandstream GXW-4004
Grandstream ATA devices
• Grandstream BT-102
• Grandstream BT-101
• Grandstream HT-286
User Manuals
Polycom
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Phones
Polycom user manual PDF
Yealink
Phones
Yealink user manual PDF
Cisco
Phones
Cisco 7940 PDF
Cisco 7960 PDF
Cisco SPA devices PDF
Linksys
Phones
Linksys SPA941 PDF
Linksys SPA942 PDF
Linksys SPA962 PDF
Sipura
Phones
Sipura SPA841 PDF
ATA Devices
Sipura SPA1000 PDF
Sipura SPA2000 PDF
Sipura SPA2002 PDF
Sipura SPA2100 PDF
Sipura SPA3000 PDF
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Aastra
Phones
Aastra 480i PDF
Aastra 53i PDF
Aastra 55i PDF
Aastra 57i PDF
Aastra 9112i PDF
Aastra 9133i PDF
Snom
Phones
Snom 320 PDF
Snom 360 PDF
Grandstream
Phones
Grandstream BT-101/BT-102 PDF
Grandstream BT-200 PDF
Grandstream GXP-2000 PDF
ATA devices
Grandstream HT-286 PDF
xten_networks
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Softphones
X-Lite PDF
X-Pro PDF
Digium
PCI cards
TDMXXX PDF
TE110P PDF
TE205P PDF
TE405P PDF
TE406P PDF
TE410P PDF
TE411P PDF
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MT 3.8.5 Dynamic Auto Provisioning
Contents
• 1 General information
o 1.1 HTTP server
o 1.2 MAC configuration files
o 1.3 Testing with curl
o 1.4 Testing with wget
o 1.5 Test HTTPS mode
o 1.6 HTTP logs
o 1.7 Syslog
• 2 Cisco Information
o 2.1 Cisco 79XX devices
o 2.2 Cisco 79X1 devices
o 2.3 Cisco SPA5xx/Cisco-Linksys SPA devices
• 3 Polycom Information
o 3.1 Provisioning method
o 3.2 Polycom devices
o 3.3 Supported features
o 3.4 General UAD config
• 4 Yealink Information
o 4.2 Yealink devices
• 5 Grandstream Devices
o 5.2 Grandstream devices
• 6 Aastra Information
o 6.2 Aastra devices
• 7 Snom Information
o 7.1 Provisioning Method
o 7.2 Snom devices
General information
In version v3.8.2 we introduced dynamic TFTP/HTTP provisioning.
Instead of writing configuration files to disk, files are now dynamically created by
PBXware HTTP application and server to device when it sends a request to the server.
NOTE:
By default (with no username and password set) HTTP provisioning is disabled.
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Setting HTTP Username and HTTP Password will automatically enable HTTP
provisioning as well. Devices that do not support Digest HTTP authentication
(username and password) cannot be auto-provisioned via HTTP, only via TFTP.
HTTP Username and HTTP Password are set under:
Settings -> Tenants -> YourTenantName (edit) -> Auto Provisioning (section) ->
HTTP Username/HTTP Password
HTTP server
HTTP application is running under following location:
http://ip-address/prov
HTTP application can handle the following scenarios (in this particular order):
• HTML/XML Directory requests
http://ip-address/prov/_directory/mac-address
• PUT requests
Only for debugging purposes, for more information check Polycom information
section.
• MAC configuration requests
Some examples of requests:

http://ip-address/prov/mac.cfg
http://ip-address/prov/cfg.mac
http://ip-address/prov/phonemac.cfg
Check Filename in each device section.
• Filesystem files (firmwares etc)
http://ip-address/prov/sip.ld
• Custom config files
http://ip-address/prov/aastra.cfg
MAC configuration files
To get configuration, devices ask for files that contain its MAC address. This
configuration requires username and password.
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Testing with curl
In order to perform testing of auto provisioning and to verify validity of content, one
needs to use curl or web browser.
More information about curl you can find here:
http://curl.haxx.se
Here is an example of command:
curl -i -u user:pass --digest http://ip-address/prov/mac.cfg
Parameters:
-i
Include HTTP protocol in output
-u
Specify username:password for authentication
--digest
We support only digest HTTP authentication, therefore this is a mandatory parameter. If

you do not specify it, curl will use Basic HTTP Authentication and it won't work.
--user-agent
This is how you can send a different User-Agent string, if you want to test HTML/XML
Directory pages.
For instance:
--user-agent "Yealink SIP-T38"
Testing in browser is advisable, if you want to test validity of large XML output.
Devices that require XML configuration will be served with text/xml Content-Type and
this will be understood by browsers. You will then know if XML document is valid or
not.
Testing with wget
In case curl is not installed (this will most often be the case on VPSes), users can use
wget to perform testing.
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Example command:
wget -S -q http://username:password@ip-address/prov/mac.cfg -O-
Parameters:
-q
Turns off verbosity and progress bar from wget.
-S
Output should include HTTP headers.
--user
HTTP username, if not specified as part of the URL.
--password
HTTP password, if not specified as part of the URL.
-O-
Output to stdout instead of file.
-U=user-agent
Use it to set User-Agent.
For more options consult:
wget --help
in your PBXware shell.
Test HTTPS mode
HTTPS also works and will work on devices that support HTTPS. Some devices do not
support self-signed certificate, therefore you will need to buy signed certificate.
curl command:
curl -i -u user:pass --digest --insecure https://ip-

address/prov/mac.cfg
Additional options:
--insecure
This options accepts self-signed certificate as valid certificate.
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HTTP response codes
Here is the list of HTTP response codes from HTTP service and its meaning:
401 Authentication Required
This header will appear when username and password is not supplied. This lets device
know it has to send username and password in next request.
200 OK
should follow after this request, in all other cases provisioning failed.
403 Forbidden
This error will appear if you do not specify valid authentication information. It will also
appear if tenant/server do not have HTTP username and password set.
404 Not Found
This error has multiple meanings but some are:
- MAC address doesn't exist
- Device doesn't exist or can't be used for auto provisioning
- Device cannot handle HTML/XML Directory
- Custom route doesn't exist
- File path is wrong
- File doesn't exist
- Request can't be handled
500 Internal Server Error
Something went horribly wrong. Usually some problematic SQL error. Please consult
Bicom Systems support.
200 OK
This happens if all is good and content is delivered to device.
HTTP logs
You can watch all devices and requests with this command (for HTTP traffic)
cd /opt/pbxware
tail -f pw/var/log/nginx/localhost.access_log
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for HTTPS:
cd /opt/pbxware
tail -f pw/var/log/nginx/localhost.ssl_access_log
Output example:
10.1.0.138 - - [23/Apr/2013:11:43:27 +0200] "GET

/prov/_directory/001565267db9 HTTP/1.0" 200 2654 "-" "Yealink SIP-T38G
38.0.23.22 00:15:65:26:7d:b9" "-"
Important bits are:
GET filename
This is what devices asked for.
200 2654
200 OK was sent, Content-Legth was 2654.
Please note, if authentication is needed, correct sequence is 401, then 200 OK.
Yealink SIP-T38G 38.0.23.22
Device User-Agent.
Syslog
Some devices can send logs via syslog.
In order to setup syslog-ng on Gentoo as syslog host, do this:
Edit /etc/syslog-ng/syslog-ng.conf
nano /etc/syslog-ng/syslog-ng.conf
and add these lines on bottom:
source net { udp(); };

destination remote { file("/var/log/remote/$FULLHOST"); };
log { source(net); destination(remote); };
Create directory /var/log/remote/
mkdir /var/log/remote
Restart syslog-ng with:
/etc/init.d/syslog-ng restart
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Cisco Information
The only provisioning method supported for Cisco 79xx devices is TFTP. HTTP
provisioning works only for Cisco SPA-Linksys devices (>= v6.x firmware).
Cisco 79XX devices
Cisco 7940, Cisco 7960 support.
Documentation: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx
Filename: SIPmac.cnf
Supported features
• Basic authentication support

• Additional UAD configuration (per extension)
• Cisco XML Directory
NOT supported:
• General UAD config
Cisco 79X1 devices
Cisco 7941, 7961 support.
Cisco 79X1 have very complicated XML output.
It is possible to provide default XML template for each device by creating template in:
/tftp/SEP_Default.xml /tftp/SEP_Device_7941.xml Template is attached in the

document.
Documentation: http://www.voip-
info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP
Firmware:
http://software.cisco.com/download/navigator.html?mdfid=269065653&flowid=5255
(requires Cisco login)
Filename
• SEPmac.cnf.xml
Content Type
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• XML format
Supported features

• Cisco XML Directory
• Voicemail access code
NOT supported

Asterisk NAT
Make sure nat=no is set in sip.conf, otherwise nothing will work.
Table 1: Cisco 79xx supported features
WARNING: Cisco 79xx devices, other than 7940 and 7960, will not be able to work
if PBXware is not in the same LAN.
These devices send UDP SIP requests from a high source port but expect a reply
on port 5060. Because of this behavior and the way NAT works, phones will always
receive response on the port their UDP SIP request was sent from and will not be
able to register. Even if devices are in the same LAN, customers will have to
manually set option NAT to No on PBXware, in Extensions -> Edit EXT ->
Network Related section, in order for Cisco 79xx devices to successfully register.
Cisco SPA5xx/Cisco-Linksys SPA devices
Cisco SPA501, 502, 504, 508, 509 525 support. Cisco-Linksys SPA941, 942, 962
support.
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IMPORTANT: These phones by default search for spa.cfg and then PBXware redirects
them to /$MA.cfg. However this only works if DHCP option 66 works properly.
HTTP
To use HTTP provisioning with authentication, username and password have to be

provided in this form:
[--uid http-username --pwd http-password] http://x.x.x.x/prov/$MA.cfg
Above works only on >=v6.x firmware, it doesn't work on firmware v5.x (Cisco-
Linksys 941...)
HTTPS
HTTPS doesn't work without special server configuration and is not supported (Cisco
signed server certificate and Client SSL certificate verification).
Filename:
• spa.cfg
• mac.cfg
Content Type:
• XML format
Supported features
All models:

• Preferred codec (first codec in list)
• Default Dialplan
• Timezone and Daylight Saving time support (via Voicemail Timezone)
• Additional UAD config (per extension)
• DHCP settings
• DNS SRV field in Servers -> Edit
• Control of max lines by internal devices list
Some models only:
BLF (With sidecart only)
• Cisco-Linksys SPA962
• Cisco SPA5XX
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Table 2: Cisco SPA supported features
General UAD config
Any content in UAD provisioning templates must be a valid XML config, and should
have start and end tag like this:
<flat-profile> <field>value</field> </flat-profile>
If XML config is invalid, it will not be included in configuration file and your phone
will not be provisioned for additional settings. Start and end tags above are not required.
Cisco firmware and files
Users have to download Cisco firmware themselves from Cisco web site.
Polycom Information
Provisioning method
Both TFTP and HTTP are supported.
HTTPS should work but requires valid certificate signed by pre-installed Certificate
Authority. It cannot be a wildcard certificate.
Polycom devices
Filenames
• MAC.cfg
Asks for where config files are, where SIP firmware is.
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• phoneMAC.cfg
Main configuration file.
• MAC-directory.cfg
BLF/Directory list.
Content Type
• XML format
Supported features
Supported

• Timezone, 12/24 clock and Daylight Saving time support (via Voicemail Timezone)
• DNS SRV field in Servers -> Edit
• BLF/Directory list
• HTTP HTML Directory
NOT supported
Table 3: Polycom supported features
General UAD config
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NOTE: General UAD config must be valid XML for that particular firmware otherwise
all configuration will fail.
PUT requests Polycom phones will send PUT requests to server and these files can be
saved on disk if upload mode is turned on.
What Polycom phones like to send:
• Log files
• Configuration override files
• Configuration saved in web interface, set on phone.
• Directory
This is not supported, as it conflicts with Enhanced Services Directory. It is not safe to
turn this feature on public internet. To turn it on, create upload directory:
cd /opt/pbxware/pw/tftp
mkdir upload
chown 101:1003 upload
It is not required to have WebDAV enabled in nginx for above to work.
For information on how to prepare your PBXware for Polycom multiple firmware
support please check our HowTo create Polycom Firmware Pack
NOTE: If you are not sure how to create Polycom firmware package please contact our
support and we will be glad to help you.
Yealink Information
Provisioning method
Either HTTP or TFTP.
HTTPS works with any certificate if "Only Accept Trusted Certificates" is Disabled.
Yealink devices
Filename
• MAC.cfg
File Format
• Format is different for T2xP and T3xP devices unless v.70 firmware is used.
Supported features
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• Codecs
• DTMF mode
• BLF/Directory list
• Timezone (no Daylight Saving Time at the moment)
• Remote HTTP XML Directory
• DNS SRV support via Servers -> Edit
Notes:
• DST is not supported

• Yealink T18 has auto provisioning bug. If provisioning server is assigned through
Option 66, manually assigned provisioning server will not work unless phone is set to
work with static IP address.
Table 4: Yealink supported features
Table 5: Yealink firmware v70 supported features
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Firmware v70
Due to Yealink omission, we are not able to detect phone firmware version, when using
TFTP. If you would like to auto provision Yealink phone with v70 firmware, please
create file v70.txt in tftp directory, like this:
cd /opt/pbxware/pw/tftp
touch v70.txt
If file v70.txt exists in /opt/pbxware/pw/tftp folder, all auto provisioning data for
Yealink phones will be in v70 format, hence all phones should be on v70 or pre-v70
version. Mixing is not supported.
Known issues
Make sure you do factory reset after upgrade to latest V70 firmware. This is known to
fix at least one problem - wrong User-Agent string for T46G device.
Wrong User-Agent:
• Yealink-T46G 28.71.0.85 28.1.0.128.0.0.0
Correct User-Agent:
• Yealink SIP-T46G 28.71.0.85 00:15:65:45:70:71
Problem with firmware older than 28.71.0.85 and T46G
Yealink T46G does not send User-Agent when requesting remote directory file, which
is required due to security reasons, hence remote directory will not work. This issue has
been fixed in 28.71.0.85 firmware.
Grandstream Devices
Provisioning method
TFTP only.
Some Grandstream devices support HTTP, but only in these two ways:
• Without username/password
• With username/password, but Basic authentication only
Therefore HTTP provisioning is not supported.
Grandstream devices
Filename
448
• cfgMAC
File Format Format is binary and is not really human-readable. It consists of

PXX=value fields.
You can pipe output to following command:
curl ... | hexdump -C -v
Supported features
Supported:

• Codecs
• DHCP
• DTMF mode (not all devices)
• Default dialplan (not all devices)
• Max lines per device limit
Not supported:
Table 6: Grandstream supported features
Aastra Information
Provisioning method
449
TFTP only, HTTP not possible as Aastra devices does not support username and
password.
Aastra devices
Filenames:
• aastra.cfg
• mac.cfg
• mac-directory.csv
Directory list in CSV format.
File Format
• Plain text
Supported features

• DHCP values
• Directory/BLF list
• Max lines per device limit
Table 7: Aastra supported features
Snom Information
Provisioning Method
TFTP only,, HTTP not possible as Snom devices doesn't support username and
password.
Snom devices
Filenames:
450
• snomXXX-mac.htm (XXX=320, 360..)
File Format
• Plain text
Supported features

• DHCP values
Table 8: Snom supported features
451
MT 3.8.5 Glossary
Glossary
Glossary of Terms
Asterisk - Implementation of PBX on whose backbone our system is built
AGI - [Asterisk Gateway Plan] - Interface through which external programs control the
dial plan
BRI [Basic Rate Interface] - ISDN configuration made out of 2 voice/data channels and
1 signaling channel
CLI [Command Line Interface] - Interface for interacting with the system using a
command line
CSV [Comma Separated Value] - File format where columns are separated by comma ','
and rows by new line
DHCP [Dynamic Hosts Configuration Protocol] - Protocol for assigning different IP

address to a device any time it connects to a network
DID [Direct Inward Dial] - Inbound line for dialing system destinations directly without
the need for an operator
DNS [Domain Name Service] - Service which translates Internet domain names into IP
addresses
DTMF [Dual Tone Multi Frequency] - A specific frequency consisting of two separate
tones sent by UAD/Phone each time a key is pressed
DynDNS [Dynamic Domain Name Service] - Service used by clients on dynamic IP

addresses which allows them to be contacted regardless of their current IP address
E.164 - Up to 15 digits international public telecommunication numbering plan. It

consists of $COUNTRY_CODE + $NATIONAL_CODE +
$SUBSCRIBER_NUMBER
IAX [Inter-Asterisk eXchange protocol] - Protocol for server-server or server-client

connection
ISDN [Integrated Services Digital Network] - Digital voice and data transmission
system over telephone wires
IVR [Interactive Voice Response] - System that manages incoming calls by playing
available options and catching user response (pressed digit)
452
LAN [Local Area Network] - Communications network limited to immediate area
LCR [Least Cost Routing] - Diverts the calls to destination via the cheapest provider
WAN [Wide Area Network] - Geographically dispersed telecommunications network
MAC [Media Access Control] - Unique code (fingerprint) assigned with network
devices
NAT [Network Address Translation] - IP packets rewriting technique used by multiple

hosts on a private network to access Internet using a single IP address
PBX [Private Branch Exchange] - A smaller version of telephone company switch
PIN [Personal Identification Number] - Four digit security code required for accessing
restricted system parts such as Voicemail, Enhanced Services, Conferences etc...
POSTFIX - Mail Transfer Agent used for routing/sending of system emails PSTN
[Public Switched Telephone Network] - The traditional, plain old telephony system
PRI [Primary Rate Interface] - ISDN configuration made out of 23/30 voice data
channels and 1 signaling channel
RTP [Real-time transport protocol] - Internet standard for transporting real-time data
(audio and video)
SIP [Session Initiated Protocol/Session Initiation Protocol] - Signaling protocol for

Internet telephony
SMTP [Simple Mail Transfer Protocol] - Protocol used to send and receive email
UAD [User Agent Device] -Telephone
TTL [Time to Live] - Time in milliseconds system will wait for the response
Zaptel - Computer telephony hardware driver
453

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Meet IP

MT 3.8.5 Getting Started

o 3.2 Administrator Details

o 3.3 Server Details (CD ROM and

o 3.7 Music On Hold

o 3.10 Logging into the system

o 3.12 License Updates

• Standard x86-compatible server hardware

• 1 GB minimum (2 GB or more recommended)

• None required. (Exception is during the installation only for)

• Free space for installation:

Local Area Networking

• Any Ethernet controller supported by the operating system

System can be administered by using one of the following web browsers:

• Internet explorer 7.0+

• Access to system server as user 'root'

CD ROM installation method is used to install the PBXware onto a commodity

• VIRTUAL SERVER installation is a PBXware system contained within a

To install the system using the CD ROM do the following:

1. Download and burn CD image from http://bicomsystems.com/download

• Before continuing please read System Requirements.

1. Download USB image from http://bicomsystems.com/download

• Before continuing please read System Requirements.

Installing PBXware in Vserver environment is only different in the beginning of the

1. Create a Vserver in SERVERware, for which you need to reference the

1. Download and extract the package to your disk

The PBXware IP address can be obtained by:

The setup wizard has a security username/password in order to prevent unauthorized

Provided email address is used as a username for logging in

Provided password is used for logging in

Server Details (CD ROM and USB version)

Re-type the Root Password entered in the field above

Select the time zone, in which PBXware is located.

The PBXware interface uses (LAN/WAN). If the PBXware is in the LAN

Whether PBXware is using DHCP or the static IP address. It is recommended to

If PBXware is in the LAN, provide its static IP address here.

This field is calculated automaticaly and contains your IP address Netmask.

If PBXware is located in LAN, set the gateway IP address here.

Server Details (Vserver version)

Server Details (Vserver version)

Re-type the Root Password entered in the field above.

The MAC address associated with the PBXware.

Select the country in which PBXware is located, ex. United Kingdom.

The number of Emergency Services in the area where PBXware is located.

• Royalty Free Music On Hold

• Repeat the wizard

• Confirm and finalize

Logging into the system

In order to update PBXware to the latest version follow these steps:

1. Login to the PBXware web interface

To upgrade your PBXware license, do as follows:

1. Login to the PBXware web interface

The applications menu provides access to all applications, site settings, SM

Data Area displays all results from various actions performed.

• System Actions Menu

Role Based Administration

Standard and Advanced Options

Full name of the user to which the device is registered

UAD/Phone extension number