Page 1
SysMaster GW 7000 Digital Gateway
User Manual
version 1.0
Page 2
Copyright © 2003 by SysMaster Corporation
All rights reserved. No part of this manual may be reproduced or transmitted in any form without
written permission from SysMaster Corporation. The information and technical data in this manual are subject to change without notice. SysMaster Corporation and its Divisions make no warranty of any kind with respect to this material, including, but not limited to, the implied warranties
of its merchantability and fitness for a given purpose. SysMaster Corporation and its Divisions
assume no responsibility for any errors that may appear in this manual and make no commitment
to update or to keep current the information in this manual.
Printed in the United States of America.
Page 3
Chapter 1 1
Contents
What is New .................. 1
Chapter 1 System Setup
System Architecture ...................................................................................... 3
Licensing ....................................................................................................... 3
General System Settings............................................................................... 4
Network Settings ........................................................................................... 5
Firewall Settings ............................................................................................ 8
System Users .............................................................................................. 10
Upgrades..................................................................................................... 11
System Setup Workflow .............................................................................. 12
Other System Management Tasks.............................................................. 15
Chapter 2 Web Console Navigation
Web Console Components ......................................................................... 17
Browsing and Managing Objects................................................................. 19
Chapter 3 Call Processing
Call Flow...................................................................................................... 21
Call Filters ................................................................................................... 30
Inbound Processing..................................................................................... 32
Outbound Processing.................................................................................. 34
Dial Groups ................................................................................................. 35
ACD (Automatic Call Distribution) ............................................................... 38
Music On Hold............................................................................................. 39
Chapter 4 Provider Setup
Overview ..................................................................................................... 41
Provider Setup............................................................................................. 43
Chapter 5 PSTN Setup
Span Configuration...................................................................................... 49
Channel Configuration................................................................................. 50
Chapter 6 Call Routing
Endpoints .................................................................................................... 53
Route Tables and Routes............................................................................ 57
SysMaster Corporation
Page 4
2
Chapter 7 RADIUS Server Setup
Overview ..................................................................................................... 61
RADIUS Server Groups .............................................................................. 61
RADIUS Servers ......................................................................................... 62
Chapter 8 IVR Setup
Introduction.................................................................................................. 65
IVR Profiles ................................................................................................. 67
IVR Objects ................................................................................................. 71
Language Servers ....................................................................................... 73
Language Server Files ................................................................................ 76
Chapter 9 PBX Server
Overview ..................................................................................................... 79
Call Filter Configuration............................................................................... 79
Inbound Profile Configuration...................................................................... 80
PBX Group Configuration............................................................................ 82
PBX Extensions........................................................................................... 84
Chapter 10 Voice Mail Server
Introduction.................................................................................................. 89
Voice Mail Groups ....................................................................................... 90
Voice Mail Boxes......................................................................................... 91
Chapter 11 Callback Server
Overview ..................................................................................................... 93
CallBack Server Setup ................................................................................ 99
Chapter 12 Conference Server
Introduction................................................................................................ 101
Conference Server .................................................................................... 102
Chapter 13 Follow-Me Server
Introduction................................................................................................ 107
Follow-Me Server ...................................................................................... 107
Chapter 14 System Monitoring
Calls Overview .......................................................................................... 111
Current Calls ............................................................................................. 111
Recent Calls .............................................................................................. 111
Gateway Monitor ....................................................................................... 113
SysMaster Corporation
Page 5
Chapter 1 3
Chapter 15 Appendix A
SysMaster Gateway Command Line Interface.......................................... 115
SysMaster Corporation
Page 6
4
SysMaster Corporation
Page 7
What is New 1
What is New
Version 02.0.47
1 SIP GW: New “Domain Name” field in Endpoints for SIP Gateways. To be used in To/
Register URI
2 SIP GW - Prevents sending empty quoted string (““) in name-addr (From) field as some
implementations do not decode it
3 SIP GW - Receive info digits (ANI-II)
Version 02.0.46
1 SIP GW: Ability to configure User Agent in System Configuration > SIP User Agent
2 SIP GW: Refuse incoming initial requests with To tag and return reason code 481
3 SIP GW: Encode message-summary body by using CRLF instead of LF
4 SIP Proxy: Forward body in NOTIFY
5 SIP Proxy: Do not set ports in From/To field
6 SIP Proxy: When changing Host in Contact the Port field will also be changed for most
request/response methods
7 SIP Proxy: Use “Expires: 0” for unsubscribe
8 SS7/MTP3: Signaling Points: State information, estimators
9 SS7/ISUP: Circuit Groups: State information, estimators
10 SS7/ISUP: Add “Called Party NOA” to Outbound Profile to allow changing the Nature
of Address Indicator in outgoing calls
11 SS7/ISUP: Add “FCI Call Type” to Outbound Profile to allow selecting between
national and international outgoing calls in Forward Call Indicators
12 PSTN: Busy detection for FXS signaling
SysMaster Corporation
Page 8
2 What is New
SysMaster Corporation
Page 9
Chapter 1 System Setup 3
Chapter 1 System Setup
System Architecture ............................................................................................................................3
Licensing...............................................................................................................................................3
General System Settings .....................................................................................................................4
Network Settings ..................................................................................................................................5
Firewall Settings...................................................................................................................................8
System Users......................................................................................................................................10
Upgrades.............................................................................................................................................11
System Setup Workflow.....................................................................................................................12
Other System Management Tasks....................................................................................................15
System Architecture
Licensing
The SysMaster gateway is built by modules. This allows for flexible adding of different
functionalities over time. You can view the currently installed modules by browsing the
Navigator tree to System Management > Licensed Modules.
SysMaster Corporation
Page 10
4 General System Settings
In the right pane of the Navigator you will see:
Module Full name of the module.
ID Coded short name.
Status Displays the current status of the module and the date
when it will expire.
Description Designated the name of the provider. The name could
contain any combination of numerical and alphabetical
characters.
Using the Navigator tree - System Management > Licensed Strings. In the right pane of the
Navigator you will see the license string of the gateway.
General System Settings
Overview
Parameters:
Gateway ID Specifies the Gateway ID. The Gateway ID could be
entered as either numerical or alphabetical value.
Domain Name Refers to the group name assigned to all computers and
network devices present on a company’s network. A
company’s domain name is usually in the form of
<companyName>.com
Host Name Defines the host name of the device. The host name is the
name used in identifying the device on a network and is
combined together with the domain name to form the
Internet address of the SysMaster device. The host name
could be comprised of any combination of alphabetical
and numerical characters in tandem with underscore,
hyphen or dot notations.
NTP Server Refers to the Network Time Protocol (NTP) Server
designed to synchronize computer clocks over a network.
The time synchronization is conducted on hourly bases
based on the time of the specified NTP Server.
Time Zone By default SysMaster operates in the Universal Time
Coordinated (UTC) zone or Greenwich Mean Time
(GMT) (same as UTC). In order for the management
company to change the default time zone, a different one
should be selected.
SysMaster Corporation
Page 11
Chapter 1 System Setup 5
DNS Server 1 Defines the DNS Server used on a gateway level by the
system for resolving computer Internet names. In the
processes Internet domain and host names are translated
to IP addresses. There could be up to two DNS Servers
defined by having there respective IP addresses
specified. The IP address should be entered as x.x.x.x
notation, where x denotes a number from 0 to 255.
The proper DNS setup enables the system to perform
upgrades over the Internet.
DNS Server 2 Defines the second DNS Server used on a gateway level
by the system for resolving computer Internet names. In
the processes Internet domain and host names are
translated to IP addresses. There could be up to two DNS
Servers defined by having there respective IP addresses
specified. The IP address should be entered as x.x.x.x
notation, where x denotes a number from 0 to 255.
address of the URL.
Backup Interval Specifies how often a backup of the gateway database
should be performed.
Managing General System Settings
To Edit General System Settings
1 From the Navigation tree, select the System Management > System Configuration
node.
2 From the right pane of the Navigator, select the first table entry.
3 From the Navigator menu, select ‘Edit Settings’.
A dialog will show up.
4 Fill in the data and click on the Apply button to enforce the settings. The settings take
effect immediately.
Network Settings
Network Configuration
A Network Configuration specifies an IP address together with network within which the
gateway is deployed. It includes additional parameters to provide information about the
networks in which the gateway resides.
SysMaster Corporation
Page 12
6 Network Settings
Network Configuration Parameters:
Local IP Address Assigns the local IP address of the network. The local IP
address will define the Local Network and servers (mail
server, web server, etc.) on the network could “listen” to
the local IP address defined.
Network Device Specifies the network device. All hosts within the
established Local Network would be available on the
network device selected.
Available choices are:
eth0
eth1
Visibility: Public / Private Determines the state of visibility of the Local Network.
Public visibility indicates that the defined Local IP
Address could be accessible via remote hosts located on
remote gateways. Private visibility indicates that the
Local IP Address would be accessible only through IP
addresses located in the Local Network). When the latter
scenario is present no gateways are necessary for routing
to take place.
Web Access Allows an access to the Web interface from remote
locations using this IP address and network device.
SSH Access Allows a Secure Shell (SSH) access through the specified
IP address and network device. Secure Shell allows a
secure remote administration of the device to be
performed.
SysMaster Corporation
Page 13
Chapter 1 System Setup 7
Managing Network Configurations
To Add Network Configuration
1 From the Navigation tree, select the System Configuration > Network Configuration
node.
2 From the Edit Menu select ‘Add Network Configuration’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to complete the operation.
To Edit Network Configuration
1 From the Navigation tree, select the System Configuration > Network Configuration
node.
2 From the right pane of the Navigator, select the Network Configuration to be edited.
3 From the Edit Menu select ‘Edit Network Configuration’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to complete the operation.
To Delete Network Configuration
1 From the Navigation tree, select the System Configuration > Network Configuration
node.
2 From the right pane of the Navigator, select the Network Configuration to be deleted.
3 From the Navigator menu select the ‘Delete Network Configuration’ to complete the
deletion.
Network Routes
Network Routes are used in accordance to their relevance to the target network. For
example, network routes are sorted in descending order as far as the target network is
concerned. Thus, the first route to be used would be /32 (host route) and the last one
0.0.0.0/0.
Network routes do not use default IP addresses. Network routes select the IP address to be
used based on the Gateway and network device assigned to the route.
Route Parameters:
Destination Network Specifies the target host that the route could be traced to.
The destination network (target host) could be assigned
as either host route (IP/32) or a network route (IP/24).
Gateway Specifies the IP address of the gateway participating in
the route. The IP address specified could be either public
or private one. If private IP address is present the
gateway would participate in direct routing.
SysMaster Corporation
Page 14
8 Firewall Settings
Network Device Specifies the network device responsible for sending
traffic from the target network to the host network.
Available choices are:
eth0
eth1
Status Indicates whether the network route is activated or not.
Managing Network Routes
To Add Network Route
1 From the Navigation tree, select the System Configuration > Network Routes node.
2 From the Edit Menu select ‘Add Network Route’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to complete the operation.
To Edit Network Configuration
1 From the Navigation tree, select the System Configuration > Network Routes node.
2 From the right pane of the Navigator, select the Network Route to be edited.
3 From the Edit Menu select ‘Edit Network Route’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to complete the operation.
To Delete Network Configuration
1 From the Navigation tree, select the System Configuration > Network Route node.
2 From the right pane of the Navigator, select the Network Route to be deleted.
3 From the Navigator menu select the ‘Delete Network Route’ to complete the deletion.
Firewall Settings
Overview
SysMaster gateway comes with an integrated Firewall. The task of the firewall is to restrict
the access to the server from the Internet for security reasons. It can also be used to eliminate
some of the common TCP/IP based attacks.
Parameters:
Source IP Network Specifies the source IP network from which packets
would be transported.
SysMaster Corporation
Page 15
Chapter 1 System Setup 9
Destination IP Network Specifies whether packets should be sent to the specified
destination formed by the Destination IP Network,
network bits and port in order to match the rule.
Action Specifies the action of the firewall filter. Available options
are:
Disabled - a firewall filter cannot be applied to
packets.
Allow - packets will be processed.
Reject - packets will not be processed an error
message will be sent in response.
Drop - packets matching the firewall filter
parameters would be ignored.
Protocol Specifies whether a specific protocol should be used in
determining which packets of data can or cannot
transport information.
Direction Specifies whether the firewall filter will be applied on
inbound or outbound packets.
Managing Firewall Filters
To Add Firewall Filter
1 From the Navigation tree, select the System Configuration > Firewall Filter node.
2 From the Edit Menu select ‘Add Firewall Filter’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to complete the operation.
To Edit Firewall Filter
1 From the Navigation tree, select the System Configuration > Firewall Filter node.
2 From the right pane of the Navigator, select the Firewall Filter to be edited.
3 From the Edit Menu select ‘Edit Firewall Filter’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to complete the operation.
To Delete Firewall Filter
1 From the Navigation tree, select the System Configuration > Firewall Filter node.
2 From the right pane of the Navigator, select the Firewall Filter to be deleted.
3 From the Navigator menu select the ‘Delete Firewall Filter’ to complete the deletion.
SysMaster Corporation
Page 16
10 System Users
System Users
Overview
System users are required so that the access to the administration console is controlled and
managed. Each user needing access to the administration console should have a system user
account. The system supports three levels of resource access and privileges enforced
through roles:
Admin role - Grants full privileges to access and modify any resource, with the
exception of accessing voicemail audio files of regular PBX users;
Guest role - Grants only viewing privileges similar to those of an administrator.
User role - Grants privileges of PBX users to view their respective voicemail boxes
and listen/delete voicemail messages (files).
Conference PBX role - Grants full privileges to Conference PBX users.
Follow-Me role - Grants full privileges to Follow-Me users.
VoiceMail role - Grants full privileges to VoiceMail users.
Parameters:
Login Name Specifies the login name of the user. The login name is
used for authentication of users and in determining their
administrative functions in front of the system.
The login name is entered in the form of a string that
could be combined by either numerical or alphabetical
(or both) characters.
Password Specifies the password of the user. The password,
together with login name are used in the process of user
authentication and authorization. Passwords should be
safeguarded. For safety reasons, it is highly
recommended that passwords are as complex as possible
and changed on frequent basis.
Please, consider the following options when choosing a
user password:
It is advisable that your password should:
Contain between 8 and 14 characters
Contain both lowercase and uppercase
alphabetical characters in addition to numbers
Not appear in any dictionary
Contain non-alphabetical characters such as !, @,
#, $, %, ^, &, *, etc
Confirm Password Requires that the password previously entered be
Full Name Designates the name of the account holder.
confirmed.
SysMaster Corporation
Page 17
Chapter 1 System Setup 11
Role Determines the role of the user to be assigned. The
following user roles are available:
Admin
VoiceMail User
PBX User
VoiceMail/PBX User
Managing System Users
To Add System User
1 From the Navigation tree, select the System Configuration > System User node.
2 From the Edit Menu select ‘Add System User’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to complete the operation.
To Edit System User
1 From the Navigation tree, select the System Configuration > System User node.
2 From the right pane of the Navigator, select the System User to be edited.
3 From the Edit Menu select ‘Edit System User’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to complete the operation.
To Delete System User
1 From the Navigation tree, select the System Configuration > System User node.
2 From the right pane of the Navigator, select the System User to be deleted.
3 From the Navigator menu select the ‘Delete System User’ to complete the deletion.
Upgrades
Overview
SysMaster Gateway provides an upgrade functionality. It allows administrators to upgrade
the SysMaster gateway to the latest version. The system allows only for upgrades.
Downgrading is not allowed.
Performing Upgrades
The upgrade procedure can be performed only by system users having an administrative
privileges (through the admin role). The procedure includes 2 steps:
1 Refresh the Upgrade List
This action makes the SysMaster gateway contact the sysmaster.com site and checks for
newer versions available for upgrade. In order for this functionality to work, please
SysMaster Corporation
Page 18
12 System Setup Workflow
make sure the SysMaster has correctly set a Network Gateway (to provide access to the
Internet) and a DNS server and that generally it has access to the Internet. The status of
the available for download upgrades is “not downloaded“.
2 Download the Upgrade file
This action selectively downloads the required upgrade. After download it is ready to
be installed. The downloaded upgrade is stored internally on the SysMaster gateway.
The downloaded upgrade has a status “not installed“.
3 Install the Upgrade File
This action applies a selected upgrade to the SysMaster gateway. The installed upgrade
has status “installed“.
NOTE: The system does not support downgrade.
To Refresh the Upgrade List
1 From the Navigation tree, select the System Configuration > Upgrades node.
2 From the Edit Menu, select ‘Refresh Upgrade List’.
3 The SysMaster gateway contacts the sysmaster.com web site and lists all upgrades
available for download.
To Download the Upgrade File
1 From the Navigation tree, select the System Configuration > Upgrades node.
2 From the list on the right pane of the Navigator, select the upgrade to be installed
3 From the Edit Menu, select ‘Download Upgrade’.
4 The SysMaster gateway will download and store the upgrade files internally.
To Install the Upgrade to the System
1 From the Navigation tree, select the System Configuration > Upgrades node.
2 From the list on the right pane of the Navigator, select the upgrade to be installed.
3 From the Edit Menu select ‘Upgrade System’.
4 The SysMaster gateway will perform the upgrade and the SysMaster gateway will have
to be restarted.
System Setup Workflow
Overview
The order of administration and setup of the SysMaster gateway is closely related to the
way the system objects and resources are allocated and the way they interact between
themselves. Below is a diagram illustrating the system object dependencies.
SysMaster Corporation
Page 19
Chapter 1 System Setup 13
SysMaster Corporation
Page 20
14 System Setup Workflow
The Setup workflow diagram shows the steps for setting up the SysMaster gateway. It
outlines 6 steps:
SysMaster Corporation
Page 21
Chapter 1 System Setup 15
1 System Setup
This section covers all basic functionality necessary for operating the gateway. It deals
with TCP/IP networking, Upgrading, System Users, Firewall filters
2 PSTN Setup
The PSTN Setup phase deals with configuring the PSTN spans and channel parameters.
PSTN Spans are used in call filters to catch PSTN originated traffic. Span/channels is
used in defining Dial Groups.
3 Provider Setup
Provider Setup deals with setting up providers as well assigning Provider rates for the
purposes of implementing Least Cost Routing algorithms when the gateway choses
from among multiple endpoints, belonging to different providers, to route the call.
Providers are attached to endpoints.
4 IVR Setup
IVR is dependent on Language servers/ Language Server files. In addition, for the
purposes of the PBX, there could be built custom audio menus that rely on IVR Objects.
IVR functionality works in tight relationship with RADIUS communication. IVR
functionality is optional.
5 RADIUS Setup
RADIUS setup includes defining first RADIUS group and then RADIUS servers for the
respective group. RADIUS servers are not always required - depending on the mode in
which the gateway works. E.g. in H.323 proxy mode, RADIUS servers are optional.
6 Call Routing, Call Processing, Endpoints Setup
At this last stage of the system setup, administrators can build the main logic for
processing calls.
Endpoints setup includes the definition of all gateways, gatekeepers, dial groups etc
endpoints with which the gateway will communicate. Use Dial Groups (through
Endpoint definitions) to terminate calls to PSTN and Spans (in Call Filters) to capture
calls from the PSTN.
Call Routing requires to first create routing tables and then populate them with routes.
Each route is a pair of area code and endpoint.
Call Processing parameters like Inbound Profiles are policies for aggreagited
processing instructions imposed on prefiltered calls. The process of call filtering is done
on the level of Call Filters. Outbound Profiles include call manipulation behavior
imposed right before the call “leaves“ the gateway.
The above described Setup workflow is suitable for setting up any operational mode of the
gateway.
Other System Management Tasks
Overview
The console allows administrators to perform the following commands on some of the basic
servers running on the gateway:
Restart VoIP Gateway
SysMaster Corporation
Page 22
16 Other System Management Tasks
Restart SMS Server - related to SysMaster Callback server
Restart DataBase Server - highly undesirable actions
Restart Mail Server - related to SysMaster Callback server
Restart All Servers
Restart SysMaster Device - restarts all servers
Halt SysMaster Device - stops all servers.
Restore Database - this command will restore the database from the most recent
database backup. Administrators can setup the frequency at which the databases
performs backup of its data through the System Configuration dialog.
All of these commands can be reached from the Start menu of the console.
IMPORTANT: Please, be careful when restarting any server!
SysMaster Corporation
Page 23
Chapter 2 Web Console Navigation 17
Chapter 2 Web Console Navigation
Web Console Components................................................................................................................17
Browsing and Managing Objects......................................................................................................19
SysMaster Web Console Client features a web-driven, browser-based intuitive and
powerful interface. The console interacts with the SysMaster administration web server for
managing the system in a client-server fashion. When starting the console, a login prompt
window will show up. A valid username and password should be provided. Usernames
and passwords are case sensitive. The system allows for only one user to login with a given
system user account at a time. The user is allowed to perform operations according to the
privileges assigned by the administrator.
Web Console Components
SysMaster Corporation
Page 24
18 Web Console Components
1 Start Button – start point for running the Navigator window, show the Help window,
Log out or close the console;
2 Task Bar – hosts all buttons hooked to opened windows of the application;
3 Start Menu – Contains basic commands for starting a new Navigator window, show the
Help window, Log out and Close the console;
4 Status Bar – Displays the currently logged user in the system;
5 Help Window – Displays general help information or context information;
6 Navigator Window – Serves as a browser of system objects for setup of the system;
All windows can be:
Moved - drag them using the title bar;
Minimized – use the minimize button on the window title bar;
Maximized – use the maximize button on the window title bar;
Closed – use the close button on the window title bar.
The menu system on a Navigator Window contains the following commands:
File
• New – Opens a new Navigator Window;
SysMaster Corporation
Page 25
Chapter 2 Web Console Navigation 19
• Print – Prints the contents of the right pane of the Navigator window;
• Close – Closes the window;
View
• Refresh – Refreshes the right pane of the Navigator window;
Edit
The Edit menu is dynamically changed, based on the selected object type from the
tree. It always offers commands that are relevant to the currently selected object.
• Add – Adds a new object;
• Modify – Modifies an object selected in the right pane of the Navigator
window;
• Delete – Deletes an object selected in the right pane of the Navigator window;
Help
• Contents – Displays the Help window;
• About Us – Shows the current version of the product;
Browsing and Managing Objects
The SysMaster Web Console features a Navigator window for browsing through the system
objects and settings.
To open the Navigator window:
1 Use Start Button > Navigator;
2 Use the tree in the left pane of the Navigator window to browse through the objects and
settings. With SysMaster every object can be listed in the right pane of the Navigator;
3 Click on a selected object from the tree;
4 On the right pane there will be listed all objects in the system relating to the object type
selected in the tree;
5 Select an object to modify or delete;
6 Use the menu commands of the Navigator window. Alternatively, use the ellipsis
button … or double click on the selected object to edit its definition.
SysMaster Corporation
Page 26
20 Browsing and Managing Objects
SysMaster Corporation
Page 27
Chapter 3 Call Processing 21
Chapter 3 Call Processing
Call Flow..............................................................................................................................................21
Call Filters ...........................................................................................................................................30
Inbound Processing...........................................................................................................................32
Outbound Processing........................................................................................................................34
Dial Groups .........................................................................................................................................35
Call Flow
PSTN to PSTN Calls
All call processing is done based on the diagram entitle “Call Flow - PSTN-to-PSTN, PSTNto-VoIP”
In this setup there could be two cases:
1 The gateway detects that the call is to be routed back through PSTN
2 The gatekeeper, based on its routing tables, decides the call should be terminated by the
same origination gateway i.e. be terminated in the PSTN
Case 1
1 The user dials into the gateway.
2 The call filter catches the call based on ANI, DNIS, or selected Span.
3 Next the gateway processes the call based on the selected Inbound Profile.
4 The gateway performs IVR processing according to the IVR Profile as indicated by the
selected Inbound Profile.
5 The gateway performs RADIUS authentication according to the selected RADIUS
Server group. This is indicated in the Inbound Profile.
SysMaster Corporation
Page 28
22 Call Flow
6 The calling party is authenticated
7 The calling party provides destination number
8 RADIUS authorization is performed
9 The gateway matches the destination number against the routing table selected by the
Inbound Profile.
10 The area code selected points to an endpoint containing a dialing group that on its part
contains PSTN span channels
11 The gateway tries to connect to the destination through the PSTN span/channels as
specified in the dialing group.
12 Upon call disconnect, the gateway sends accounting message to the RADIUS server
chosen from the selected RADIUS server group.
Case 2
1 The user dials into the gateway.
2 The call filter catches the call based on ANI, DNIS, or selected Span.
3 Next the gateway processes the call based on the selected Inbound Profile.
4 The gateway performs IVR processing according to the IVR Profile as indicated by the
selected Inbound Profile.
5 The gateway performs RADIUS authentication according to the selected RADIUS
Server group. This is indicated in the Inbound Profile.
6 The calling party is authenticated
7 The calling party provides destination number
8 RADIUS authorization is performed
9 The gateway matches the destination number against the routing table selected by the
Inbound Profile.
10 The area code selected, points to a gatekeeper
11 RAS signalling to the gatekeeper is initiated. The Gateway should be registered with the
Gatekeeper before they can communicate with each other.
SysMaster Corporation
Page 29
Chapter 3 Call Processing 23
12 The gatekeeper finds in its routing tables that this call should be terminated by the
origination gateway.
13 The gateway creates an incoming call as if coming from its gatekeeper.
14 A second call filter catches the call through an Endpoint filter (there should be the local
gatekeeper selected)
15 The Inbound Profile should point to an IVR Profile with CDR only policy, no RADIUS
Server group, and Routing Table pointing to PSTN (i.e. Endpoint of type Dial Group).
16 The call is routed to a PSTN span/channel from the selected Dial Group.
17 Connection is established to the remote PSTN peer.
18 Upon call disconnect, the gateway sends accounting CDR message to the RADIUS
server (chosen from the selected RADIUS server group of the first Call Filter/Inbound
Profile).
SysMaster Corporation
Page 30
24 Call Flow
SysMaster Corporation
Page 31
Chapter 3 Call Processing 25
PSTN to VoIP Calls
All call processing is done based on the diagram entitle “Call Flow - PSTN-to-PSTN, PSTNto-VoIP”.
This is a standard setup for VoIP calls.
1 The user dials into the gateway.
2 The call filter catches the call based on ANI, DNIS, or selected Span.
3 Next the gateway processes the call based on the selected Inbound Profile.
4 The gateway performs IVR processing according to the IVR Profile as indicated by the
selected Inbound Profile.
5 The gateway performs RADIUS authentication according to the selected RADIUS
Server group. This is indicated in the Inbound Profile.
6 The calling party is authenticated.
7 The calling party provides destination number.
8 RADIUS authorization is performed.
9 The gateway matches the destination number against the routing table selected by the
Inbound Profile.
10 The area code selected, points to a gatekeeper
11 RAS signalling to the gatekeeper is initiated. The Gateway should be registered with the
Gatekeeper before they can communicate with each other.
12 The gatekeeper finds in its routing tables that this call should be terminated by a
termination gateway.
13 The gatekeeper returns to the Origination Gateway the IP address of the termination
gateway in an ACF message (as part of the RAS signaling).
14 The gateway initiates a call setup to the remote gateway H.323 peer.
15 The termination gateway connects to the destination number through PSTN.
SysMaster Corporation
Page 32
26 Call Flow
16 The connection between both ends is established
17 After call is disconnected, the gateway sends accounting message to the RADIUS with
CDR records of the call.
VoIP to PSTN calls
This setup is relevant when the gateway is a termination endpoint. This is the case when
other parties want to terminate calls to area codes serviced by the gateway through PSTN.
1 An incoming call request is received into the Sysmaster gateway.
2 The call filter catches the call based on source number prefixes or by endpoint.
3 The Gateway selects the Incoming Profile as indicated in the call filter.
4 The IVR profile should be CDR Only indicating no audio services are provided.
5 The RADIUS server selected will be used only for accounting purposes.
6 The gateway checks the selected Routing Table and selects a Dial Group Endpoint
7 SysMaster gateway select channels to connect through the PSTN.
8 The gateway connects to the destination number.
9 Upon call disconnect, the gateway sends accounting message to the RADIUS server and
sends CDR data for the call.
SysMaster Corporation
Page 33
Chapter 3 Call Processing 27
VoIP to VoIP calls
There are three cases possible for this setup:
Case A. IVR/RADIUS over IP
Case B. H.323 Proxy Mode
Case C. Termination to VoIP phone or VoIP software PC client
Case A. IVR/RADIUS over IP
In this scenario, there is a phone connecting to analog gateway. The gateway does not have
neither gatekeeper, nor IVR server to contact, nor RADIUS server for AAA. The analog
SysMaster Corporation
Page 34
28 Call Flow
gateway acts like a transparent proxy between the end user and the gateway. This scenario
allows for building a cheap infrastructure across large areas with low volume of calls.
1 The user dials into the analog gateway.
2 The analog gateway contacts the main SysMaster digital gateway over the Internet.
3 An incoming call request is received into the Sysmaster gateway.
4 Next the gateway processes the call based on the selected Inbound Profile.
5 The gateway performs IVR processing according to the IVR Profile as indicated by the
selected Inbound Profile. All IVR messages are sent over IP and are played by the
analog gateway to the user.
6 The gateway performs RADIUS authentication according to the selected RADIUS
Server group. This is indicated in the Inbound Profile.
7 The calling party is authenticated based on PIN.
8 The calling party provides destination number.
9 RADIUS authorization is performed.
10 The gateway matches the destination number against the routing table selected by the
Inbound Profile.
11 The selected area code points to a gatekeeper
12 RAS signalling to the gatekeeper is initiated. The Gateway should be registered with the
Gatekeeper before they can communicate with each other.
13 The gatekeeper finds in its routing tables that this call should be terminated by a
termination gateway.
14 The gatekeeper returns to the Main Gateway the IP address of the termination gateway
in an ACF message (as part of the RAS signaling).
15 The gateway initiates a call setup to the remote gateway H.323 peer.
16 The termination gateway connects to the destination number through PSTN.
17 The connection between both ends is established.
18 Upon call disconnect, the Main Gateway sends accounting message to the RADIUS with
CDR records of the call.
Case B. H.323 Proxy Mode
1 An incoming call request is done.
2 The call filter catches the call based on source number prefixes or by endpoint.
3 The Gateway selects the Incoming Profile as indicated in the call filter.
4 The IVR profile should be Pass-Through indicating no audio services are provided.
5 The RADIUS server selected will be used only for accounting only or AAA purposes
(optional).
6 The gateway checks the selected Routing Table and selects the gateway as endpoint to
which the outgoing call will be established. It is not allowed to use H.323 Gatekeeper as
endpoint. In the case of H.323 proxy, there is no codec translation performed. The
SysMaster Corporation
Page 35
Chapter 3 Call Processing 29
gateway simply redirects all packets of H225, H245, RTP. It can optionally redirect
inbound VoIP IVR messages too.
7 After call disconnect, the proxy may send accounting message to the RADIUS server if
such one is specified.
Case C. Termination to VoIP phone or VoIP software PC client
1 An incoming call request is received into the Sysmaster gateway.
2 The call filter catches the call based on source number prefixes or by endpoint.
3 The Gateway selects the Incoming Profile as indicated in the call filter.
4 The IVR profile should be CDR Only indicating no audio services are provided.
5 The RADIUS server selected will be used only for accounting purposes.
6 The gateway checks the selected Routing Table and selects the gatekeeper as an
Endpoint
7 SysMaster gateway performs RAS signaling asking for permission to connect to the
destination number as well as receive the destination IP address of the destination
H.323 peer.
8 The gatekeeper detects that there is an endpoint registered for the destination number.
SysMaster Corporation
Page 36
30 Call Filters
9 The gatekeeper returns to the SysMaster gateway the IP address of the destination
H.323 peer in an ACF message (as part of the RAS signaling).
10 The gateway initiates a call setup to the destination H.323 peer (that can be a VoIP
phone or a VoIP software PC client).
11 The connection between both ends is established
12 After call is disconnected, the gateway sends accounting message to the RADIUS with
CDR records of the call.
Call Filters
Overview
Call Filters are used on only incoming calls and only before an answer from the call has been
received. Call Filters could be viewed as firewalls granting or denying access to remote
callers. Incoming calls are authenticated based on ANI or DNIS authentication types, and
once the authentication parameters match the call filter, it can accept or reject the incoming
call. If incoming calls come from the Internet (e.g. from gateways, gatekeepers, VoIP
phones), ANI becomes Source Number, and DNIS becomes Destination Number.
When an incoming call is accepted, the incoming call will be associated with an Inbound
Profile for further processing. The inbound call will be processed according to the IVR
Profile, Routing Table, RADIUS Server Group, or PBX Group as allocated to it in the
Inbound Profile.
Parameters:
Filter Name Name of the filter.
Priority Grants a level of priority to the created call filter. The
priority can range from 0 to 20. The higher number
means higher priority. Priority is used to determine
which Call filter will match and process an incoming call
when there are two or more matching Call filters.
In case more than one Call Filter matches an incoming
call, the system tries to resolve which Call filter will take
over the call processing by checking the following
parameters in the order they appear:
Higher priority
Longer DNIS
Longer ANI
Endpoint Designated an endpoint to the call filter. The filter rules
are applied to the endpoints matching this endpoint. The
matching is done based on an IP address.
SysMaster Corporation
Page 37
Chapter 3 Call Processing 31
Mask (0-32) Specifies the mask of the IP address to be matched. For
example mask 24 will enforce matching of the first 3
octets of the IP address of the Endpoint. This way,
multiple gateways originating from the same network
can match the rules of the filter.
Span Defines the Span to be matched. The only way to catch
PSTN incoming calls is to specify the span they come
from. If there are four spans and the system must catch
calls from all the spans, there should be a seperate filter
for each span. PSTN calls cannot be catched using Dial
Groups as an endpoint
ANI Prefix Specifies that ANI (Automatic Number Identification)
Prefix (for calls coming from PSTN) or a Source Number
(for calls coming from VoIP channels) prefix that will be
used to match incoming calls.
DNIS Prefix Specifies that DNIS (Dialed Number Identification
Service) Prefix (for calls coming from PSTN) or a
Destination Number (for calls coming from VoIP
channels) prefix that will be used to match incoming
calls.
Inbound Profile Assigns an Inbound profile to the call filter. The Inbound
profile is used to further process the call providing IVR,
RADIUS, and routing functionality.
Description Allows a short description (up to 71 characters) of the
Call Filter to be entered.
Advanced Rules Advanced Rules allow for flexible enforcement of the call
filter based on time or endpoint availability.
Managing Call Filters
Call Filters should be generally created last following the setting up all other prerequisite
resources. The workflow for managing Call Filters is the following:
1 Create the necessary Endpoint, if Endpoint matching will be performed.
2 If Span matching will be performed, make sure all Spans are correctly defined. You can
match calls coming from the PSTN lines only through Spans. Should administrators
want to match all PSTN traffic, independent of which Span it comes, they should create
as many Call Filters as the number of Spans - one Span per Call Filter. All such Call
Filter may share the same Inbound Profile processing. Beside this, within a Call Filter
administrators can specify even a separate channels, participating in the Call Filter. This
provides for finer granularity in differentiating the origin of incoming calls and their
subsecuent processing.
3 Prepare the required Inbound Profile to process the matching call. To this end, there
should be an appropriate IVR Profile, RADIUS Groups, Routing Table, PBX Group.
Some of these may not be required to define an Inbound Profile.
SysMaster Corporation
Page 38
32 Inbound Processing
To Add Call Filter
1 From the Navigation tree, select the Call Processing > Call Filters node.
2 From the Edit Menu select ‘Add Call Filter’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the modifications.
4 In the case, Span filtering is selected - to match calls originating from the PSTN -
administrators will have to further edit the definition of the Call Filter. There they will
be able to specify which span channels will participate in the call filtering.
To Edit Call Filter
1 From the Navigation tree, select the Call Processing > Call Filters node.
2 From the right pane of the Navigator, select the Call Filter to be edited.
3 From the Edit Menu select ‘Edit Call Filter’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to enforce the modifications.
To Delete Call Filter
1 From the Navigation tree, select the Call Processing > Call Filters node.
2 From the right pane of the Navigator, select the Call Filter to be deleted.
3 From the Edit Menu, select the Delete menu item.
Inbound Processing
Overview
Inbound Profile specifies a set of processing parameters that will be associated with an
incoming call. These include the following:
ANI/DNIS (Sorce/Destination Number) translation
IVR Profile
Routing Table
RADIUS Server Group
PBX Group
Once an inbound profile for an incoming call has been created, it could be added to the
overall call processing by assigning it to a call filter.
DNIS/ANI Translation
DNIS/ANI translation is perofmed in the Inbound processing first. It is necessary so that
the supplied numbers are prepared for correct routing. The ANI/DNIS number would be
rewritten according to the pattern specified. Each pattern has the form of leftPart =
rightPart. The left part is matched against the prefix of the destination number. Each
marked text is then replaced with the corresponding text from the right part of the field. The
SysMaster Corporation
Page 39
Chapter 3 Call Processing 33
right part contains list of texts separated with colon (:). If the Left part does not have text
marked with [] then the whole left part is marked for replacement.
Examples:
Pattern: '123=456' Number: '123789': new number: '456789'
Pattern: '1[(1:2)]3=456' Number: '123789': new number: '14563789'
Pattern: '1[(1:2)]3S=456' Number: '123789': new number: '14563789'
Pattern: '1[X]3[N]S=456' Number: '123789': new number: '1456389'
Pattern: '@##[123][]X[]=444:55:6' Number: '5555#123789': new number: '5555#444557689'
Pattern: '[]123=444' Number: '123789': new number: '444123789'
Parameters:
DNIS Translation Defines a DNIS pattern of translations.
ANI Translation Defines an ANI pattern of translations.
IVR Profile Assigns and IVR Profile to the inbound profile of the
incoming call. System administrators could select an
appropriate IVR Profile from the list of available ones.
Routing Table Associates the inbound profile with a particular already
created Routing Table.
RADIUS Group Specifies the RADIUS Group the inbound profile would
belong to.
PBX Group Specifies the PBX Group the inbound profile would
belong to.
Description Allows a short description (up to 71 characters) of the
Inbound Profile profile to be entered.
Managing Inbound Profiles
The workflow for creating Inbound Profile is the following:
1 Plan the appropriate ANI/DNIS number translations. This is necessary for correct
routing of the call.
2 Specify an appropriate IVR Profile. In the case, an IVR Profile needs to be created, make
sure all Language Servers and language files are in place.
3 Specify an appropriate Routing Table to route the call. If such table does not exist, make
create one. For this, you will need to have a provider with provider rates (that serve
only for the Least Cost Routing algorithms, and not for billing) as well as Endpoints that
will terminate the calls.
SysMaster Corporation
Page 40
34 Outbound Processing
4 Specify a RADIUS Group. To this end the system must have configured RADIUS server
groups and RADIUS servers in them. You will have to consider what kind of RADIUS
services the system will need. i.e. AAA (Authentication, Authorization, Accounting) or
Accounting only.
5 Specify a PBX Group. This is necessary if you plan use this Inbound Profile for PBX call
processing and routing.
To Add Inbound Profile
1 From the Navigation tree, select the Call Processing > Inbound Profile node.
2 From the Edit Menu select ‘Add Inbound Profile’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the modifications.
To Edit Inbound Profile
1 From the Navigation tree, select the Call Processing > Inbound Profile node.
2 From the right pane of the Navigator, select the Inbound Profile to be edited.
3 From the Edit Menu select ‘Edit Inbound Profile’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to enforce the modifications.
To Delete Inbound Profile
1 From the Navigation tree, select the Call Processing > Inbound Profile node.
2 From the right pane of the Navigator, select the Inbound Profile to be deleted.
3 From the Edit Menu, select the Delete menu item.
Outbound Processing
Overview
Outbound Profiles are used by Endpoints before an outgoing call has been established.
Creating an Outbound Profile ensures that the endpoint(s) associated with it would be able
to translate Caller IDs and Destination Numbers according to the requirements. Endpoints
and their respective Outbound Profiles are used in the routing process.
Parameters:
ANI Translation Defines an ANI pattern of translations. The destination
number would be rewritten according to the pattern
specified.
DNIS Translation Defines a DNIS pattern of translations. The destination
number would be rewritten according to the pattern
specified.
SysMaster Corporation
Page 41
Chapter 3 Call Processing 35
PBX Group Specifies the PBX Group the outbound profile would
belong to.
Description Allows a short description (up to 71 characters) of the
Inbound Profile profile to be entered.
Managing Outbound Profiles
The workflow for creating Outbound Profile is the following:
1 Plan the appropriate ANI/DNIS number translations. They determine how these
parameters will send to the call terminating parties.
2 In the case of PBX processing and routing, specify a PBX Profile.
To Add Outbound Profile
1 From the Navigation tree, select the Call Processing > Outbound Profile node.
2 From the Edit Menu select ‘Add Outbound Profile’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the modifications.
To Edit Inbound Profile
1 From the Navigation tree, select the Call Processing > Outbound Profile node.
2 From the right pane of the Navigator, select the Inbound Profile to be edited.
3 From the Edit Menu select ‘Edit Outbound Profile’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to enforce the modifications.
To Delete Inbound Profile
1 From the Navigation tree, select the Call Processing > Outbound Profile node.
2 From the right pane of the Navigator, select the Inbound Profile to be deleted.
3 From the Edit Menu, select the Delete menu item.
Dial Groups
Overview
Dial Groups represent groups of Endpoints and PSTN Span/channels. Dial Groups creation
is in two stages. In the first stage, they are simply created as objects and at the second stage
Endpoints are added to them. Each Dial Group has an ID which is used to attach them to an
Endpoint. E.g. an Endpoint of type Dial Group can have a local IP address 127.0.0.1:10
where 10 is the ID of the selected Dial Group attached to the endpoint.
Each Dial Group has a Hunt Type determining the way endpoints are accessed in the group.
It can be one the following:
SysMaster Corporation
Page 42
36 Dial Groups
Sequential/Asc - the routing result is filled in the order in which the endpoints are
defined in the group.
Sequential/Desc - the routing result is filled in the reverse order in which the
endpoints are defined in the group.
First Available - Dial Groups of such Hunt Type can have only Span Channels
elements (no other endpoints). The gateway selects the first available span channel
and attempts a call. If it fails, it doesn’t attempt to route the call through another
channel.
Broadcast - all endpoints in the group are connected simultaneously. If the number
specified in the max result is exceeded, no more connections are tried to be
established.
Rotation - acts similarly to the Sequential order but with each call that is terminated
in the dial group, the previously top endpoint in the routing result order goes to the
bottom of the list.
Parameters:
Dial Group Name Defines the name of the created dial group. The name
could contain any combination of alphabetical and
numerical characters.
Hunt Type Specifies the method according which endpoints would
be served by the dial group.
Available options are:
Sequential/Asc
Routing result is always obtained from the first
endpoints in the dial group.
Sequential/Desc
Routing result is always obtained by the last
endpoints in the dial group
First Available
Broadcast
All endpoints in the dial group are present in the
routing result. The amount of endpoints present
could be controlled by the value of the “Max
Results” field.
Rotation
Endpoints are chosen in a manner identical to
Sequential/Asc. The difference being that every
other routing result will be obtained by the next
endpoint in the dial group.
Max Results (1-500) Defines how many endpoints of a dial group could be
present in the routing result. The routing result is an
ordered list of Endpoints to which the gateway attempts
to route a call.
SysMaster Corporation
Page 43
Chapter 3 Call Processing 37
Connect Timeout (1-120) Denotes the period of time (in seconds) beyond which
dial group connection attempts will be terminated. The
dial group connect timeout period could be in the range
of 1 to 120 seconds
ACQ Profile Specifies the ACQ Profile assigned to the dial group.
Managing Dial Groups
The workflow for creating Dial Groups is the following:
1 Plan the appropriate Hunt Type. If you plan to use the gateway to terminate to PSTN,
then the Rotating algorithm is most suitable.
2 Select a proper Max Results number. You may want to select smaller number up to the
number of call lines. Max Result is responsible for the length of the list of possible lines/
endpoints ready to be used to put the call through.
3 Select an Appropriate ACD Profile. ACD (Automatic Call Distribution) Profiles are
suitable for PBX call processing.
4 Plan the Endpoints, Spans and Channels to be used in this Dial Group. Nesting of Dial
Groups is not allowed.
To Add Dial Gr ou p
1 From the Navigation tree, select the Call Processing > Dial Group node.
2 From the Edit Menu select ‘Add Dial Group’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the modifications.
4 To specify the spans and channels assigned to a Dial Group proceed with Editing the
Dial Group once it is created.
To Edit Dial Group
1 From the Navigation tree, select the Call Processing > Dial Group node.
2 From the right pane of the Navigator, select the Dial Group to be edited.
3 From the Edit Menu select ‘Edit Dial Group’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to enforce the modifications.
To Delete Dial Group
1 From the Navigation tree, select the Call Processing > Dial Group node.
2 From the right pane of the Navigator, select the Dial Group to be deleted.
3 From the Edit Menu, select the Delete menu item.
SysMaster Corporation
Page 44
38 ACD (Automatic Call Distribution)
ACD (Automatic Call Distribution)
Overview
ACD is a module allowing administrators to define various behaviors(profiles) for call
distribution within a dial group. Each Dial Group can be bound to only one ACD profile.
Attaching small number of manageable ACD profiles to a large number of Dial Groups,
allows for easy readjustment of the ACD rules for multiple Dial Groups at a time.
Parameters:
Name Defines the name of an ACD profile.
Music On Hold Specifies the name of the music-on-hold file to be played
while calls are in queue.
Calls to Start Queueing Specifies the number of concurrent calls to a dial group
after which the gateway will put calls in queue.
Generally, this number should be the number of
endpoints within a dial group to which the ACD profile
is attached.
Calls to Refuse Queueing Specifies the number of concurrent calls to a dial group
Greeting Iverval (sec) Indicates the time interval at which greeting message is
Auto Retry Call Interval (sec) Specifies the interval at which queued calls attempt to
Managing ACD Profiles
To Add ACD Profile
1 From the Navigation tree, select the ACD > ACD Profiles node.
2 From the Edit Menu select ‘Add ACD Profile’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the modifications.
4 To specify the spans and channels assigned to a ACD Profile proceed with editing the
selected ACD Profile once it is created.
To Edit ACD Profile
1 From the Navigation tree, select the ACD > ACD Profiles node.
after which the gateway will drop incoming calls.
played.
connect.
2 From the right pane of the Navigator, select the ACD Profile to be edited.
3 From the Edit Menu select ‘Edit ACD Profile’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to enforce the modifications.
SysMaster Corporation
Page 45
Chapter 3 Call Processing 39
To Delete ACD Profile
1 From the Navigation tree, select the ACD > ACD Profiles node.
2 From the right pane of the Navigator, select the ACD Profile to be deleted.
3 From the Edit Menu, select the Delete menu item.
Music On Hold
Overview
The Music On Hold module controls the music played to callers while waiting in queue. The
module allows administrators to create multiple Music-on-hold profiles for modular
assignment to ACD profiles. Each Music-on-hold profile contains multiple files that are
played to the waiting callers.
Music-On-Hold Profiles
Music-on-hold profiles can only be created and deleted. Modifying is not possible.
Parameters:
Name Defines the name of an Music-On-Hold profile. The
Managing Music-On-Hold Profiles
To Add Music-on-hold Profile
1 From the Navigation tree, select the Music On Hold > Music-On-Hold Profiles node.
2 From the Edit Menu select ‘Add Profile’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to complete the creation.
To Delete Music-on-hold Profile
1 From the Navigation tree, select the Music On Hold > Music-On-Hold Profiles node.
2 From the right pane of the Navigator, select the Delete Profile to be deleted.
3 From the Edit Menu, select the Delete menu item.
Music-On-Hold Files
profile is listed in the tree for adding Musc-on-hold files.
Music-on-hold files are the actual music files that are played. The accepted audio format is
wave, au, gsm. where PCM - uLaw 8kHz - is for “.au” extension and GSM for “.gsm”
extension.
SysMaster Corporation
Page 46
40 Music On Hold
Parameters:
New File Name Defines the name of an Music-On-Hold file.
Managing Music-On-Hold Files
To Add Music-on-hold File
1 From the Navigation tree, select the Music On Hold > Music-On-Hold Files node.
2 From the Edit Menu select ‘Upload File’.
A dialog box shows up.
3 Click on the “Browse” button and locate the file to be uploaded.
4 Click on the Process File to actully upload the file.
To Rename Music-on-hold File
1 From the Navigation tree, select the Music On Hold > Music-On-Hold Files node.
2 From the right pane of the Navigator, select the file to be renamed.
3 From the Edit Menu, select the Rename File menu item.
To Delete Music-on-hold File
1 From the Navigation tree, select the Music On Hold > Music-On-Hold Files node.
2 From the right pane of the Navigator, select the Delete File to be deleted.
3 From the Edit Menu, select the Delete menu item.
SysMaster Corporation
Page 47
Chapter 4 Provider Setup 41
Chapter 4 Provider Setup
Overview .............................................................................................................................................41
Provider Setup....................................................................................................................................43
Overview
A provider is a telco company that provides its VoIP equipment and infrastructure to the
Management company to terminate VoIP calls made through the gateways/gatekeepers of
the Management company. The provider company can provide either gateways or
gatekeepers to terminate calls. The SysMaster digital gateway uses provider definitions to:
1 Implement route failover.
2 Implement least cost routing of calls based on an algorithm for selecting the least
expensive route.
3 Collect CDR records for calls through all providers.
To accomplish the tasks listed above, the system introduces the following objects:
Provider - a telco company providing VoIP termination services;
Provider rate - a rate definition assigned to a area code location.
In order for the system to use the provider rates data, the system allows you to attach a
provider an endpoint as well as attach provider to a route. This way the system can access
information about providers and their rates and take call routing decisions based on the
routing rules. For more information please refer to the routing section.
Provider
Parameters:
Provider Name Designated the name of the provider. The name could
contain any combination of numerical and alphabetical
characters.
Provider Rates
Parameters:
Provider Designated the name of the provider.
SysMaster Corporation
Page 48
42 Overview
Location Denotes the name of the location the newly created rate
would apply to. In most instances, a location is referred
as the geographical area encompassing a single or a
number of area codes.
Area Code Assigns the unique area code the newly created rate
would apply to. The area code should correspondent to
the location entered above.
Init Time (sec) Specifies the duration (in seconds) of the initial time
period. The initial time period (interval) starts from the
moment when a call is placed and ends based on the
value entered in the Init Time field.
Init Charge (cents) Allocates the amount (in US cents) that would be charged
during the initial time period (Init Time) of the call.
Sample Time (sec) Specifies the duration of the Sample Time (interval)
period. The Sample Time period represents at what
duration the call would be billed. If a call is to be billed on
per minute bases, the Sample Interval must be 60 sec.
If no Sample Time is entered, the system will assign a
default Sample Time value of 0.
Sample Charge (cents) Specifies the amount (in US cents) that would be charged
on per Sample Time basis.
NOTE: Sample Charge charges are not charges per
second, but charges per Sample Time. To calculate per
second call charges Sample Rate should be divided by
Sample Interval.
Sample Charge / Sample Time = per second charges
Increment Time (sec) Specifies the value of the time increment to be applied to
the base duration of the call. The Increment Time value
would represent a single time unit that would be used in
segmenting the call for billing purposes.
For instance: If the Increment Time is specified as 10 sec
and a user initiates a call for 31 sec, the call would be
segmented into 4 billing time units. As a result the user
would NOT be charged for the 31 sec of call time but for
40 sec call time.
Call Origination Charge (cents) Specifies the amount (in US cents) imposed on the
management company by the Call Origination Provider.
Call Origination Provider is network provider offering
services at the point of call origination.
Call origination charges represent an expense to the
management company.
SysMaster Corporation
Page 49
Chapter 4 Provider Setup 43
Call Termination Charge (cents) Specifies the amount (in US cents) imposed on the
management company by the Call Termination Provider.
Call Termination Provider is network provider offering
services at the point of call termination.
Call termination charges represent an expense to the
management company.
The figure below visually explains the meaning of the rate attributes.
Provider Setup
Provider Management
To Add a Provider
1 From the Navigation tree, select the Provider Management > Provider node.
2 From the Edit Menu select ‘Add Provider’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the provider creation. During
provider creation, the system assigns an unique ID of the provider for further reference.
To Edit a Provider
NOTE: Changes made to currently existing provider definitions will propagate throughout
the entire system i.e. endpoint, provider rate, area code route definitions.
1 From the Navigation tree, select the Provider Management > Provider node.
2 Select the node to be edited and from the Edit Menu select ‘Edit Provider’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the changes.
SysMaster Corporation
Page 50
44 Provider Setup
To Delete a Provider
NOTE: Deleting a provider will automatically delete a all relating provider rates, endpoints
defined for the provider, as well as area code routes for the provider.
1 From the Navigation tree, select the Provider Management > Provider node.
2 Select the node to be deleted and from the Edit Menu select ‘Delete Provider’.
3 The delete command will be completed and the list with providers refreshed.
Provider Rate Management
Provider rates are defined per provider. The system will logically allow you to create
provider rates only for existing providers. As mentioned earlier, provider rates do not
participate in call billing but only to hint the system how to intelligently route calls based on
least call routing. Rates are defined per area code. The area code may be simply the first
number of an area code, not necessary the full area code to a destination.
You can add provider rates in two ways: either through the add dialog, or from the Import
command.
To Add a Provider Rate through the Add Rate Dialog
1 From the Navigation tree, select the Provider Management > Provider Rates > [a
specific provider] node.
2 From the Edit Menu select ‘Add Provider Rate’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the provider rate creation.
Please, pay special attention to the area code data you fill in.
In order to use the Import dialog, to import rates you have to comply with the following
comma separated data format:
"Country", "Area Code", "Init Time", "Init Charge", "Sample Time",
"Sample Charge", "Increment Time", "Call Origination Charge", "Call
Termination Charge"
Example:
usa, 1, 10, 60, 1, 10, 0, 2, 2
can, 2, 10, 60, 1, 10, 0, 2, 2
To Add Provider Rates through the Import Dialog
1 From the Navigation tree, select the Provider Management > Provider Rates > [a
specific provider] node.
2 From the Edit Menu select ‘Import Provider Rate’.
A dialog box shows up.
3 Specify the local file and click on the Apply button to import the rates.
SysMaster Corporation
Page 51
Chapter 4 Provider Setup 45
TIP: The Provider Rates node in the tree lists all providers as tree nodes. There is a
bracketed number indicating system ID of the respective provider.
To Edit a Provider Rate
1 From the Navigation tree, select the Provider Management > Provider Rates > [a
specific provider] node.
2 From the list of rates in the right pane, select the rate to edit. From the Edit Menu select
‘Edit Provider Rate’. A dialog box shows up.
3 Edit the settings to be changed and click on the Apply button to enforce the changes.
TIP: The provider rates list in the right pane of the Navigation window has a tab named ‘Is
Modified’ that indicates whether a rate has been modified once it has been created (either
through the Add dialog or through Rates import operation).
To Delete a Provider Rate
1 From the Navigation tree, select the Provider Management > Provider Rates > [a
specific provider] node.
2 From the list of rates in the right pane, select the rate to edit. From the Edit Menu select
‘Delete Provider Rate’.
3 The delete command will be completed and the list with provider rates will be
refreshed.
To Delete All Provider Rates
1 From the Navigation tree, select the Provider Management > Provider Rates > [a
specific provider] node.
2 From the Edit Menu, select ‘Delete All Provider Rates’.
3 The delete command will be completed and the list with provider rates will be emptied.
To Export All Provider Rates of a Provider
1 From the Navigation tree, select the Provider Management > Provider Rates > [a
specific provider] node.
2 From the Edit Menu select ‘Export Provider Rates’.
3 A popup window will prompt you to download a text file containing all provider rates
in a comma separated format. The format is the same as the one used to import rates.
Provider Call History
The SysMaster gateway allows the administrators to view detailed call records for calls that
have been terminated by the respective provider. Furthermore, administrators can export
CDRs for billing purposes or additional further data processing.
To Export All CDRs for a Provider
1 From the Navigation tree, select the Provider Management > Provider Call History >
[a specific provider] node.
SysMaster Corporation
Page 52
46 Provider Setup
2 From the tree, select a provider node to view its calling history in the right pane of the
Navigator.
3 From the Menu, select the ‘Export Call History’ menu. A popup window will prompt
you to save the data to your local computer. The data is exported in comma-separated
text format.
The SysMaster gateway interface provides a convenient tool to interactively view call
history using relevant time and area code filters.
To View Call History
1 From the Navigation tree, select the Provider Management > Provider node.
2 A list of all providers will appear in the right pane of the navigator. Select the provider
for which call history data will be viewed.
3 From the Menu, select the ‘Call History’ menu. A new window will show up. In this
window, you can view monthly and daily call history records for calls that have been
terminated by the respective provider.
To View Call History by Time
1 Click on the month/day link to drill down to CDR data for the month/day selected.
2 At each stage of drilling down you will see aggregated data for the selected period
(month or day)
3 Use the navigational helpers to navigate to the desired level of time. These links are
located at the top of the table.
To View Call History Filtered By Area Code
1 The Navigate to the desired level for which the filter will work (e.g. specific day, month
etc.)
2 Hit the search button to show all calls that match the filter criteria.
The Call Detail Records have the following parameters:
Calls/Successful Specifies the successful to failed call ratio. This is for all
calls that has been routed through the selected provider.
Average Latency(sec) The average latency for the duration of the call.
Average Successful Rate
Specifies the successful to failed calls ratio.
(ASR)(%)
Originated Specifies the number of call that have been originated
from the provider.
Answered Specifies the number of calls that have been terminated
(answered) by the provider.
SysMaster Corporation
Page 53
Chapter 4 Provider Setup 47
Average Cost Specifies the average cost for the calls that have been
routed through the selected provider. The price is
calculated based on the provider rates for the respective
provider.
Call Time Calling time for the duration of the routed calls.
Incoming Bytes Specifies the inbound VoIP data volume in bytes with
respect to the SysMaster gateway for the duration of the
calls.
Outgoing Bytes Specifies the outbound VoIP data volume in bytes with
respect to the SysMaster gateway for the duration of the
calls.
Incoming Packets Specifies the inbound VoIP data volume in IP packets
with respect to the SysMaster gateway for the duration of
the calls.
Outgoing Packets Specifies the outbound VoIP data volume in IP packets
with respect to the SysMaster gateway for the duration of
the calls.
SysMaster Corporation
Page 54
48 Provider Setup
SysMaster Corporation
Page 55
Chapter 5 PSTN Setup 49
Chapter 5 PSTN Setup
Span Configuration ............................................................................................................................49
Channel Configuration.......................................................................................................................50
From the Edit Menu, select the Delete menu item. .........................................................................52
Span Configuration
OverviewRADIUS
Spans represent trunks (spans) of T1/E1/ISDN digital PSTN lines. The SysMaster gateway
supports T1/E1/ISDN lines according to the installed voice card.
T1 Spans usually support 23+1 channels, i.e. 24 channels in CAS mode and 23+1 channels in
ISDN mode (one D channel). The channel allocation is as follows: CAS uses 24 voice
channels; ISDN uses 23 voice channels (channel 24 is used for D channel).
E1 Spans support 31 channels in CAS mode and 30 B channels and one D channel in ISDN
mode. The channel allocation is as follows: CAS uses 30 voice channels (1-15,17-31 for voice
and 16 for signaling).
ISDN uses 30 voice channels (channel 16 is used for D channel).
Spans can be used in:
Call Filters to filter incoming calls from the PSTN lines
Dial Groups, and this way they can participate indirectly in Endpoints. Endpoints
can be used in Routing tables for outgoing call routing. This is why, attaching spans
and channels to Dial Groups is the only way to route calls to the PSTN side.
The SysMaster gateway automatically detects the settings of the T1/E1 PSTN card that is
installed on the device.
Parameters:
Span ID Specifies the sequence number (ID) of the PSTN span.
Supported numbers are:
Span ID: #1
Span ID: #2
Span ID: #3
Span ID: #4
SysMaster Corporation
Page 56
50 Channel Configuration
Span Status: Enabled /
Disabled
Signalling Specifies the signaling type of the digital voice line. CAS
Span Name Assigns a name to the created Span.
Span Management
To Add Span Configuration
1 From the Navigation tree, select the PSTN/TDM Management > Span Configuration
node.
2 From the Edit Menu select ‘Add Span Configuration’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the modifications.
If enabled, the channels within the span can be used,
otherwise they will be blocked.
(Channel Associate Signaling): used for channeling the
service. Each channel represents a phone line. The
signaling data is sent on the same data channel.
ISDN PRI there are two options for signaling:
CPE (Customer Premises Equipment)
Network Side
To Edit Span Configuration
1 From the Navigation tree, select the PSTN/TDM Management > Span Configuration
node.
2 From the right pane of the Navigator, select the Span Configuration to be edited.
3 From the Edit Menu select ‘Edit Span Configuration’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to enforce the modifications.
To Delete Span Configuration
1 From the Navigation tree, select the PSTN/TDM Management > Span Configuration
node.
2 From the right pane of the Navigator, select the Span Configuration to be deleted.
3 From the Edit Menu, select the Delete menu item.
Channel Configuration
Overview
A Channel represents a bandwidth allocation within a Span enough to accomodate one call
leg. A single call between two parties requires one channel for each party.
There are two kinds of channel signaling:
SysMaster Corporation
Page 57
Chapter 5 PSTN Setup 51
Loop Start and Ground Start use Dial Tones for start, and only DTMF signaling
(Dual Tone Multi-Frequency).
E & M (Ear and Mouth) use Immediate or Wink methods for start, and MF (Multi-
Frequency), or DTMF (Dual Tone Multi-Frequency) signaling.
Loop Start and Ground Start do not provide DNIS (Dialed Number Identification Service,
called party number) and ANI (Automatic Number identification, the calling party number)
address information. Such address information can be sent from the telephone company by
using other methods such as E&M.
Loop Start is the most used signaling for analog and digital lines. On T1 lines the signaling
uses bits A and B. The method is susceptible to glare and should not be used on bidirectional lines where both ends can place calls simultaneously.
Ground Start is similar to Loop Start but provides immediate seizure indication to avoid the
glare.
The Loop Start and Ground Start signaling can be provided to different parties:
Foreign Exchange Office (FXO) signaling: this signaling is provided to FXS devices
(inside phones or PBXs systems).
Foreign Exchange Station (FXS) signaling: this signaling is provided to CO lines or
outside PBXs.
There are other methods that provide two-way calling, all based on E&M:
E&M Wink Start: the caller waits for wink (the called party should go off-hook for
short period of time). Then the caller sends the address information.
E&M Immediate: the caller does not wait for wink before sending address
information.
There are extensions to the E&M signaling:
E&M Feature Group B (MF): this method uses Multi-Frequency and can provide
DNIS address information.
E&M Feature Group D (MF): this method uses Multi-Frequency and can provide
ANI and DNIS address information.
E&M Feature Group D (DTMF): this method uses Dual Tone Multi-Frequency and
can provide ANI and DNIS address information.
Parameters:
Channes: Available / Assigned Specifies the available channels to be assigned to the
channel configuration setting.
Signal Type Allows a signal type to be selected. Used when span
signaling is set to CAS.
SysMaster Corporation
Page 58
52 Channel Configuration
Tx Gain (-20dB:20dB) Specifies the value for the Tx Gain. The Tx Gain is used to
amplify the transmitted voice level. The Tx Gain could
only be adjusted in the range of -20.0 - +20.0 db.
Rx Gain (-20dB:20dB) Specifies the value for the Rx Gain. The Rx Gain is used to
amplify the received voice level. The Rx Gain could only
be adjusted in the range of -20.0 - +20.0 db.
Accept Caller ID Specifies whether the Caller ID for an incoming call to be
accepted or not.
Caller ID Specifies Caller ID to use for incoming calls. It is used
only for FXO signaling. If the number of the Caller ID
ends with '+' sign, it will be replaced with a channel
number.
For example:
- Assigned Caller ID: 1234, Resulting Caller ID: 1234
- Assigned Caller ID: 1234+, Resulting Caller ID for
channel 24: 123424
Time Rule
Channel Management
To Add Channel Configuration
1 From the Navigation tree, select the PSTN/TDM Management > Channel
Configuration > [Span] node. All nodes under channel configurations represent the
spans that are active.
2 From the Edit Menu select ‘Add Channel Configuration’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the modifications.
To Edit Channel Configuration
1 From the Navigation tree, select the PSTN/TDM Management > Channel
Configuration > [Span] node.
2 From the right pane of the Navigator, select the Span Configuration to be edited.
3 From the Edit Menu select ‘Edit Channel Configuration’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to enforce the modifications.
To Delete Channel Configuration
1 From the Navigation tree, select the PSTN/TDM Management > Channel
Configuration > [Span] node.
2 From the right pane of the Navigator, select the Channel Configuration to be deleted.
3 From the Edit Menu, select the Delete menu item.
SysMaster Corporation
Page 59
Chapter 6 Call Routing 53
Chapter 6 Call Routing
Endpoints............................................................................................................................................53
Route Tables and Routes...................................................................................................................57
Endpoints
Overview
An endpoint is associated with the party to which a call setup could be made or with the
party from which calls could be received by the Management Company. In accordance to
the direction of the call flow an endpoint could be either Incoming (a call has been placed
towards us) or Outgoing (a call has been placed from us).
Each endpoint created refers to a designated Provider. When output routing is performed
the desired endpoint is determined on the basis of its associated Provider.
NOTE: Endpoints of type H.323 should be assigned a status of active/local listener when
created.
Types of endpoints
H.323 Gateway
H.323 Gateway is an entity that communicates using the H.323 set of protocols. It translates
PSTN packets into IP packets and vice versa. The former is called encoding and the latter decoding. H323 Gateways are used by H.323 gateway or by H.323 Proxy for incoming or
outgoing calls. If a gatekeeper is specified as a controller/gatekeeper, then the gateway uses
the gatekeeper to send ARQ (Admission Request) message over the RAS channel to it. The
gateway waits for Request Confirm or Request Reject messages.
H.323 Gatekeeper
H.323 Gatekeepers can be used as outgoing endpoints as well as RAS (Registration,
Admission and Status) channel for incoming and outgoing calls of the H.323 gateway.
If a Gatekeeper is specified then the gateway can receive calls from its IP, without
performing RAS signaling for it.
When a Gatekeeper is used as an endpoint, it can redirect a call to a remote H.323 endpoint
but the system needs to know only about the gatekeeper itself. In addition, the gatekeeper
can perform number translation, bandwidth control as well as other tasks related to call
processing.
SysMaster Corporation
Page 60
54 Endpoints
When a Gatekeeper is specified as a controller for H.323 gateway, all incoming and
outgoing calls, the gateway use RAS signalling through it. In this way the Gateway binds
itself to the Gatekeeper.
H.323 Proxy
H.323 is a special mode of work of the H.323 gateway where it simply redirects VoIP
messages. It does not change the encoding of the messages. It can use RADIUS processing
optionally as well as use a special type of IVR (pass-through) instructions. Authentication is
done via tech-prefixes. This mode can be used as a tool to hide the source of your voice
traffic.
By default, H.323 Gateway listens at all IP addresses at port 1720 or at specified Gateway
Listen IP addresses. If a gateway uses both as a H.323 gateway and H.323 proxy, there could
be port contention because H.323 uses port 1720 too. This is why H.323 Proxy and H.323
Gateway should not have common Listen IP addresses.
On the other hand it is recommended that Listen IP addresses are specified in order to
escape common resource interference. When using the gateway as both H.323 Gateway and
H.323 Proxy, the system check procedures stop services with no specific Listen IP address
assigned (as the lack of a specific Listen IP address indicates the gateway will listen at all
available IP addresses).
Callback Server
The Callback server endpoint denotes the callback functionality of the SysMaster gateway.
It can work in several modes based on the authentication mode as well as the way it is
requested - web request, email, SMS message or the way authentication is done - ANI, PIN,
DID.
SIP Gateways
SIP Gateways are similar to H.323 Gateways with the main difference that they can specify a
Default Source number to be used in outgoing calls. This IP address should be present in the
list of IP addresses of the SIP gateway or to be a Local IP address in the cases when the
Listen IP address is 0.0.0.0 (i.e. all IP addresses).
SIP Registrars
SIP Registrars are used for registration of the SIP Gateway ID together with its IP address in
a registry. This way the gateway is known to other SIP Peers through its registered name.
All such registration is being refreshed on a timely bases which is useful if the SIP Gateway
host changes its IP address.
Dial Group
Dial Groups represent aggregated entity of other endpoints or PSTN spans/channels. A
Dial Group cannot contain other dial groups.
SysMaster Corporation
Page 61
Chapter 6 Call Routing 55
Call Drop Endpoint
This type of endpoint is used to drop calls. It can be used in building blocking call filters for
dropping calls with certain parameters such as ANI, DNIS, SPAN, ENDPOINT.
Parameters:
Name Name of the endpoint. The name could consist of any
combination of alphabetical or numeric characters.
IP Specifies the IP address of the endpoint.
Typ e Specifies what type of the endpoint.
Status Indicates whether the endpoint is active or disabled. An
endpoint could also be related to a local or remote IP
address by selecting the active/local listener status.
GK/SIP Controller Assigning a GK/SIP Controller to an endpoint ensures
that the endpoint would be able to facilitate RAS
messaging.
Number of Ports (1-1000) Defines the number of ports available to the endpoint.
The number of ports represents the maximum number of
calls allowed to the endpoint.
Priority (0-20) Assigns a level of priority to the endpoint. The higher the
number the higher priority would be assigned to the
endpoint. A higher level of priority of an endpoint means
that the endpoint will have a better chance participating
in the route of the outgoing call.
Description Allows a short description (up to 71 characters) of the
endpoint to be entered.
Provider Name Designates the name of the provider the endpoint will be
associated with.
Outbound Profile Specifies the outbound profile the endpoint will be
associated with.
Country Specifies the country where the endpoint is located.
Min Digits (0-10) Specifies the minimum destination number digits in
order for the system to put a call through this endpoint.
Available Codecs Specifies the codecs that are supported. Administrators
can chose only from the listed set of codecs. The assigned
codecs should be listed in order of their preference.
Jitter (20-10000 msec) Specifies the Jitter Buffer selection. Jitter is used to buffer
incoming packets to allow for high latency networks to
support VoIP communication. The recommended
settings are between 50 and 300 ms jitter buffer.
SysMaster Corporation
Page 62
56 Endpoints
Timeout (1-120) For Gatekeepers only. Specifies the timeout between
retransmissions of the RAS (Registration, Admission and
Status - H.225.0) messages.
Default Source IP Specifies the source IP address for the endpoint. Used
only for SIP gateways.
Fast Start If selected, Fast Start VoIP negotiation would be enabled
and a faster call setup would be performed. Make sure
the endpoint needs to support this feature.
H.245 Tunneling The feature allows H.245 messages the use the existing
H.225 Transmission Control Protocol (TCP). Make sure
the endpoint needs to support this feature.
Silence Suppression Designates that the endpoint supports Silence
Suppression capable VoIP voice processing. Make sure
the endpoint needs to support this feature.
Endpoints Management
To Add Endpoint
1 From the Navigation tree, select the Route Management > Endpoint Configuration
node.
2 From the Edit Menu select ‘Add Endpoint’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the modifications.
To Edit Endpoint
1 From the Navigation tree, select the Route Management > Endpoint Configuration
node.
2 From the right pane of the Navigator, select the Endpoint to be edited.
3 From the Edit Menu select ‘Edit Endpoint’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to enforce the modifications.
To Delete Endpoint
1 From the Navigation tree, select the Route Management > Endpoint Configuration
node.
2 From the right pane of the Navigator, select the Endpoint to be deleted.
3 From the Edit Menu, select the Delete menu item.
SysMaster Corporation
Page 63
Chapter 6 Call Routing 57
Route Tables and Routes
Overview
Routing Tables are used to determine to what Endpoints an incoming call can proceed as an
outgoing call. Routing tables represent a lit of pairs of a phone number prefix against an
endpoint thus indicating a route. Routing tables are assigned from an Inbound profile and
in its turn, an Inbound profile is selected from a call filter.
The VoIP routing searches for the destination number only in the selected by the Inbound
profile Routing table.
When the gateway performs call routing it internally builds a list of endpoints through
which the call will be routed. The top endpoint of the list is attempted first. If it fails, the
next one is attempted, and so on until the call is finally routed. The gateway takes into
account the following factors when building its internal routing list for a call:
If an endpoint has bigger number for priority, then it is preferred;
Next in the list come the endpoints with lesser cost. The cost is taken from the
provider rates for the provider assigned to the Endpoint.
The result represent Endpoints and Dial Groups (wrapped in Endpoints) through which the
call should be redirected.
For each Endpoint from this list an Outbound Profile is used according to the one attached
in the respective Endpoint definition.
Routing Table Definition Parameters:
Routing Table Name of the routing table.
Enable Auto Failover Autofailover allows the routes to use numerous
endpoints for termination provided such endpoints are
available for the same route.
Timeout (1-300 sec) Denotes the period of time (in seconds) beyond which
routing query will be terminated. The routing timeout
period could range from 1 to 300 seconds.
The format of the routing tables is as follows:
"Area Code", "Endpoint IP", "Provider ID", "Cost", "Preferred"
Routes Management
The workflow for managing routes is the following:
1 Create a provider.
2 Define Endpoints that should belong only to one provider.
SysMaster Corporation
Page 64
58 Route Tables and Routes
3 Add Area code rates. The area codes may be simply prefixes (e.g. 3, 5, 22, 454 etc.).
Adding rates for Area Code prefixes determines available prefixes a provider covers
4 Create Routing (Routing Table) by assigning which area codes will be in the table
To Add Route Table Definition (Route)
1 From the Navigation tree, select the Route Management > Routes node.
2 From the Edit Menu select ‘Add Route’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the modifications.
To Edit Route Table Definition (Route)
1 From the Navigation tree, select the Route Management > Routes node.
2 From the right pane of the Navigator, select the Route to be edited.
3 From the Edit Menu select ‘Edit Route’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to enforce the modifications.
To Delete Route Table Definition (Route)
1 From the Navigation tree, select the Route Management > Route node.
2 From the right pane of the Navigator, select the Route to be deleted.
3 From the Edit Menu, select the Delete menu item.
To Import Route Tables
1 From the Navigation tree, select the Route Management > Route Table > [Route
Table] node.
2 From the tree, select the Route Table to import data into.
3 From the Edit Menu select ‘Import Routes’.
A dialog box shows up.
4 Hit on the Browse button to locate the file to import from. The files should be in the
following comma delimited format:
"Area Code", "Endpoint IP", "Provider ID", "Cost", "Preferred"
5 When ready, Click on the Import button to complete the procedure.
To Export Route Tables
1 From the Navigation tree, select the Route Management > Route Table > [Route
Table] node.
2 From the tree, select the Route Table from which data will be exported.
3 From the Edit Menu, select the Export menu item.
4 A dialog will prompt you to save the file. CLick on the Save button to save the file.
SysMaster Corporation
Page 65
Chapter 6 Call Routing 59
To Delete a Single Route or All Routes
1 From the Navigation tree, select the Route Management > Route Table > [Route
Table] node.
2 From the right pane of the Navigator, select the Route Table entry to be deleted (for
single deletion).
3 From the Edit Menu select ‘Delete Route’ - for single deletion, or ‘Delete All Routes’ for
deleting all entries.
SysMaster Corporation
Page 66
60 Route Tables and Routes
SysMaster Corporation
Page 67
Chapter 7 RADIUS Server Setup 61
Chapter 7 RADIUS Server Setup
Overview .............................................................................................................................................61
RADIUS Server Groups......................................................................................................................61
RADIUS Servers .................................................................................................................................62
Overview
SysMaster Gateway needs a RADIUS (Remote Authentication Dial In User Services) server
to authenticate and authorize call parties and at the end of to send CDR data for accounting
(billing) purposes. The RADIUS server adds a layer of security on the layer of voice
communication service provided. The SysMaster Gateway complies with the RADIUS
protocol and can communicate with any server conforming to the standards.
For the purposes of authentication and authorization, the SysMaster Gateway can work
with only one RADIUS server per call. On the other hand, for the purposes of accounting
the gateway can contact multiple RADIUS servers simultaneously.
The gateway represents a NAS (Network Access Server) client to the RADIUS server. The
NAS communicates with the RADIUS server through a password - called shared secret.
The administrators of the system should attach a RADIUS server group to each Inbound
Profile. The Inbound Profile is used to assign to prefiltered incoming calls a RADIUS Group,
IVR Profile, Routing Table, PBX group (if a PBX is installed).
RADIUS Server Groups
Introduction
The SysMaster Gateway can communicate with multiple RADIUS servers simultaneously.
In order to separate RADIUS servers for each defined provider, the system uses RADIUS
Server Groups to group RADIUS servers responsible for a provider. Within a group there
can be multiple RADIUS servers all of which can handle accounting requests
simultaneously. Within a server group there can be only one server handling authentication
requests together with authorization.
Parameters:
Name Specifies the name of the server group.
Timeout (1-60 sec) Defines the period of time (in seconds) for timeout of the
server query. RADIUS Server Group timeout could range
from 1 to 60 seconds.
SysMaster Corporation
Page 68
62 RADIUS Servers
Retry Count (1-10) Defines the number of times one query will be submitted
to the RADIUS server. Once the number of retries is over
then the system will fail-back to the next RADIUS server
in the RADIUS Group.
NAS IP Address Specifies the IP address of the Network Access Server
(which is the SysMaster gateway) from which it will
communicate with the RADIUS server. The system
allows you to select from the IP addresses that are
defined for the gateway. The gateway and the RADIUS
server preferably should reside in the same network.
Description Allows a short description (up to 71 characters) of the
conference profile to be entered.
RADIUS Server Group Management
To Add a RADIUS Server Group
1 From the Navigation tree, select the RADIUS Management > RADIUS Server Group
node.
2 From the Edit Menu select ‘Add RADIUS Server Group’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the Server Group creation.
To Edit a RADIUS Server Group
1 From the Navigation tree, select the RADIUS Management > RADIUS Server Group
node.
2 From the right pane of the Navigator, select the Server Group to be edited.
3 From the Edit Menu select ‘Edit RADIUS Server Group’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to enforce the Server Group modification.
To Delete a RADIUS Server Group
1 From the Navigation tree, select the RADIUS Management > RADIUS Server Group
node.
2 From the right pane of the Navigator, select the Server Group to be deleted.
3 From the Edit Menu select ‘Delete RADIUS Server Group’.
RADIUS Servers
Introduction
RADIUS servers are defined with the gateway so that the gateway knows how where to
authenticate, authorize and send accounting information for the calls that go through it.
RADIUS servers are defined within server groups. Within a Server Group, only one
SysMaster Corporation
Page 69
Chapter 7 RADIUS Server Setup 63
RADIUS server can be responsible for authentication/authorization while all other RADIUS
servers defined within the group should be for accounting only.
Parameters:
Name Specifies the name of the RADIUS server.
Typ e Specifies the type of the RADIUS server.
Available options are:
Disabled
This option disables the server within the server
group it is defined.
AAA
This option specifies the server will be used for
authentication, authorization and accounting.
Accounting
A RADIUS server of this type, should be used
only for accounting. The gateway uses it to send
CDR records at the end of every call. Before the
CDR records are sent, a login authentication
request is sent.
Accounting RFC
This option indicates that the server will perform
similar task as a server of type Accounting. The
difference is that there is no login authentication
procedure before the gateway sends the CDR
records.
IP Address Specifies the IP Address of the RADIUS server.
Shared Secret Specifies password through which the gateway device
Port Specifies the port at which the RADIUS server listens at.
RADIUS Server Management
To Add a RADIUS Server
1 From the Navigation tree, select the RADIUS Management > RADIUS Servers >
[RADIUS Server] node.
2 From the Edit Menu select ‘Add RADIUS Server’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the RADIUS Server creation.
will authenticate itself and initiate a communication
session with the RADIUS server.
Usually the port number is 1812/1813.
SysMaster Corporation
Page 70
64 RADIUS Servers
To Edit a RADIUS Server
1 From the Navigation tree, select the RADIUS Management > RADIUS Servers >
[RADIUS Server] node.
2 From the right pane of the Navigator, select the RADIUS Server to be edited.
3 From the Edit Menu select ‘Edit RADIUS Server’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to enforce the RADIUS Server
modification.
To Delete a RADIUS Server
1 From the Navigation tree, select the RADIUS Management > RADIUS Servers >
[RADIUS Server] node.
2 From the right pane of the Navigator, select the RADIUS Server to be deleted.
3 From the Edit Menu select ‘Delete RADIUS Server’.
To Delete a All RADIUS Servers
1 From the Navigation tree, select the RADIUS Management > RADIUS Servers >
[RADIUS Server] node.
2 From the Edit Menu select ‘Delete All RADIUS Servers’.
SysMaster Corporation
Page 71
Chapter 8 IVR Setup 65
Chapter 8 IVR Setup
Introduction ........................................................................................................................................65
IVR Profiles .........................................................................................................................................67
IVR Objects .........................................................................................................................................71
Language Servers ..............................................................................................................................73
Language Server Files .......................................................................................................................76
Introduction
The SysMaster Gateway features a flexible IVR (Interactive Voice Response) functionality
providing audio services to calling parties. The IVR module works closely with the RADIUS
server authentication, authorization and accounting. The IVR functionality features:
ANI/PIN Based Scripted Paths
SysMaster gateway provides predefined scripted paths for playing messages when the user
dials in the gateway. Each call is first authenticated by ANI, and in the case that ANI
authentication fails the, IVR Profile would be authenticated by PIN. Full audio services for
PSTN and VoIP calls are supported, as well as post-paid and pre-paid accounts are fully
supported. Through the interface, the SysMaster gateway can be programmed to only
authenticate only by ANI or only by or fall back to both.
Pass-Through Functionality
SysMaster provides special Pass-through authentication based on tech prefixes. This mode
is often used in tandem switching and PSTN switching. It can also be successfully used to
switch VoIP to VoIP calls (2 IP legs are supported in one call). No audio services are
supported. Post-paid and pre-paid accounts are fully supported;
Callback Functionality
SysMaster provides special IVR message paths to be played in connection with the way the
user is authenticated/authorized.
Flexible DTMF Routing
The gateway allows for fine management of the DTMF message routing. It can selectively
pass inband or out-of-band PCM16 frames. This can be suitable for cases when
implementing Intelligent proxy or for peers supporting only certain type of DTMF routing.
The system can enforce auto detection and intelligently select the right mode when no
specific DTMF routing is required.
SysMaster Corporation
Page 72
66 Introduction
Language Servers
The SysMaster gateway allows for defining multiple language servers. Each server specifies
an IVR server with a specific directory where audio file from the same language are located.
This way administrators can easily switch between multiple languages. All language files
should be first locally cached. This is done through the “Download Language
Server“command, executed on each server.
Multiple Language Support
The IVR module allows administrators to give their users to select between two languages
with an option. The selection message menu is played after initial dial into the gateway.
Multisession IVR
The SysMaster gateway can support multiple calls from an user that has already been
authenticated by the system. In this way, the user can place easily multiple consecutive calls
within one calling session. In order to this, the user simply has to press a predefined DTMF
- usually this is “#”.
Customer Service Redirect
This feature enables end users to quickly contact customer service representatives to discuss
call issues or buy calling time over the phone. The system provides announcements to guide
the users easily call through.
Calling Time/Balance Announcement
The IVR module can play the current balance of the user after he/she authenticates with the
system. It can also selectively play the available calling time after the user has entered the
destination number.
Pre-paid and Post-paid Account Support
All of the gateway features concerning user interaction are supported for both pre-paid and
post-paid accounts.
Low Balance Announcement
The system can be configured to play a waning message when the balance is below a certain
threshold.
IVR over IP
The SysMaster digital gateway can be used as a central point for RADIUS authentication/
authorization for IVR services to multiple analog gateways that do not have this
functionality. All IVR messages are encoded over IP (using one of the selected endpoint
codecs) and are transmitted to the remote analog gateway peer. On the other hand the
analog gateway converts the announcements back to analog signal and plays them to the
user.
SysMaster Corporation
Page 73
Chapter 8 IVR Setup 67
Auto Attendant PBX Menu System
For the purposes of building a PBX IVR system, administrators can use the SysMaster
gateway capabilities to construct arbitrarily deep levels of IVR menus. To this end, the
gateway uses IVR Profiles and IVR objects which provide great flexibility. For more
information, please refer to the IVR Objects section.
Intelligent Proxy
The Intelligent Proxy allows the VoIP Gateway to relay audio frames between H323 and SIP
channels without using codec operations (encode/decode). This way, the gateway can
achieve high call throughput. If the Intelligent Proxy does not operate, the gateway will
always decode and encode the audio frames.
If Intelligent Proxy mode operates it can be enabled for any direction (receive or transmit)
according to the following rules:
the used codecs have CPU intensive encode/decode operations
same audio codecs are used in both call legs. For this to work, SysMaster changes its
codec preference order for outgoing calls and uses the peer's preference order for
incoming calls. For H323 the Intelligent Proxy codec negotiations work better when
Fast Start is used.
if the used IVR Type is Pass-through Authentication or CDR Collection the
Intelligent Proxy does not depend on the Inband DTMF Detection requirements.
IVR Profiles
Overview
Beside IVR information, IVR Profiles hold logic that determines the way IVR is played as
well as the way authentication/authorization is done.
IVR Profiles are used for each Incoming Call. They are attached through the Inbound
Profiles which are attached to an incoming call through a Call Filter.
When an IVR profile takes control over an incoming calls, the task of the VoIP Gateway is to
perform authentication and authorization of the call using the selected RADIUS server.
Next, the gateway has to play various IVR messages to the calling party, to determine the
destination number, to call the VoIP routing to determine the endpoint to be used for the
outgoing call according to the destination number and to bridge the audio traffic between
the incomming call and the outgoing call.
Sometimes the IVR Profile does not instruct the gateway to play audio messages to the
calling party. In this case, the gateway may directly connect both parties with no messgaes
inbetween. This may the case when working in Proxy mode or in passthrough mode for
PSTN-to-PSTN call termination.
SysMaster Corporation
Page 74
68 IVR Profiles
Parameters:
Name Specifies the name of the RADIUS server.
Typ e Defines the type of the IVR Profile. IVR Profile types fully
support VoIP switching (a call can come via IP and then
be terminated via IP), as well as PSTN origination/
termination.
Available IVR Profiles types are:
ANI– designates that each call will be authenticated via
ANI (Caller ID) type authentication.
PIN -designates that each call will be authenticated via a
PIN number.
PIN / ANI - designates that each call will be first
authenticated first ANI, and in the case that ANI
authentication fails the IVR Profile would be
authenticated by PIN. Full audio services for PSTN and
VoIP calls are supported, as well as post-paid and prepaid accounts are fully supported;
CDR Only – designates that each call will be processed
and routed without authentication. Final CDR records
will be collected only. When CDR authentication is
selected, no audio services will be supported. Post-paid
and pre-paid accounts are fully supported;
Pass-Thru – designates that each call will be
authenticated via tech-prefix and then automatically
routed.
Two-Stage - each call will be processed and routed
without authentication. RADIUS Server can be used for
authorization if there is RADIUS Server Group specified;
CallBack - each call will be rejected and data passed to
Callback Server as a Call Back request.
PBX - each call will be processed and routed without
authentication. If there is RADIUS Server Group
specified it will be used for CDR Collection.
Inband DTMF Detection Specifies the status of the inband DTMF Detection to be
used of PCM16 frames.
Available options are:
Yes – enables inband DTMF detection for PCM16 frames.
No – disables inband DTMF detection for PCM16 frames.
The option could be used when the peers send out of
band DTMF tones or when the Intelligent Proxy is
enabled.
Auto/Both - detects inband and out of band DTMF
Auto/OutOfBand - accepts only out of band DTMF.
SysMaster Corporation
Page 75
Chapter 8 IVR Setup 69
Enable Intelligent Proxy Specifies whether the Intelligent Proxy option would be
activated or not. The Intelligent Proxy allows the VoIP
Gateway to relay audio frames between H323 and SIP
channels without using codec operations (encode/
decode).
Enable DTMF Relay To Leg 1 Allows for Leg 1 to receive DTMF tones. By default all
DTMF tones received on Leg 2 are relayed to Leg 1.
Enable DTMF Relay To Leg 2 Allows for Leg 2 to receive DTMF tones. By default all
DTMF tones received on Leg 1 are relayed to Leg 2.
Generate Artificial Ring Tones When checked, artificial ring tones will be generated by
the IVR profile.
Generate Connect Tone When checked, connect tones will be generated by the
IVR profile.
Language Server Specifies an applicable Language Server. All files would
be downloaded from the specified Language Server and
would be played based on the IVR Selection.
Second Language Server Specifies a Second Language Server only used if multi-
language IVR is desired.
Enable Language Selection Language Selection enables multi-lingual support. If
checked, the gateway will play a prompt for language
selection. The language files used would be the ones from
the specified two language servers.
Enable Multisession IVR Enables multi-session support. Multi-session allows
multiple sequential calls from one account. (e.g. once the
first call disconnects, the customer can initiate a second
call, and so on).
Enable Customer Service
Redirect
Enables Customer Service Redirect to the customer
service phone number (listed below). The redirect can be
done from any prompt level and will automatically
connect the call based on the dial plan.
Customer Service Number Specifies the customer service number that will be
announced by the system. If the customer service number
is dialed the caller will automatically be connected to the
customer service representative.
Fist Digit Timeout (1-30 sec) Specifies the time in seconds before the system times out
and plays an error prompt message. The period of first
digit timeout could be in the range from 1 to 30 seconds.
Digit Timeout (1-30 sec) Specifies the time in seconds before the system times out
and plays an error prompt message. The specified value
could be in the range from 1 to 30 seconds.
SysMaster Corporation
Page 76
70 IVR Profiles
PIN Length (1-20) Specifies the maximum length of the PIN number. The
PIN length could be specified in the range from 1 to 20
characters.
PIN Terminator Specifies the PIN termination character. If the caller
presses '#', or '*', or if the PIN timeout is reached, the PIN
will be submitted. PIN terminator can be either an
explicit character (#,*), or PIN entry timeout.
PIN Retries (1-5) Specifies the maximum number of PIN entry retries that
are allowed by the system. After the maximum number
of attempts is reached the system will disconnect the
caller.
Max Destination Number
Length (1-40)
Specifies the maximum length of the called station
number. This is the destination number that will be
entered by the caller.
Destination Number Terminator Specifies the Destination Number termination character.
If the caller presses '#', '*', or if Destination Number
timeout is reached, the Destination Number will be
submitted. Destination number terminator can be either
an explicit character (#,*) or destination number entry
timeout.
Destination Disconnect Specifies the character that allows multi-session call
disconnect. Once this character is pressed, the current call
will be disconnected and the system will prompt the
caller for another call. This can be either an explicit
character ( #,*) or timeout.
Destination Number Retries
(1-5)
Specifies the maximum number of Destination Number
entry retries that are allowed by the system. After the
number of attempts is reached the system will disconnect
the caller.
Enable Number Redial When checked, number redial will be performed.
Enable Pre-Paid Credit
Enables credit announcement for pre-paid callers.
Message
Enable Post-Paid Credit
Message
Enable Pre-Paid Time Message Enables credit time announcement for pre-paid callers
Enable Post-Paid Time
Message
Enable Post-Paid Timer Enables post-paid call timer (time based disconnect) for
Enables credit announcement for post-paid callers.
Enables credit time announcement for post-paid callers.
post-paid callers.
SysMaster Corporation
Page 77
Chapter 8 IVR Setup 71
Drop Message Time
(0, 10-60 sec)
End Message Enables the end of call (good bye) message
Enable Recharge Account Enables recharging of the account.
Recharge Threshold Specifies the minimum balance threshold. If the balance
Managing IVR Profiles
To Add IVR Profile
1 From the Navigation tree, select the IVR Management > IVR Profiles node.
2 From the Edit Menu select ‘Add IVR Profile’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the modifications.
Specifies the time before a warning disconnect message is
played to the caller. If number entered is greater than 0
the message is enabled. If number specified is equal to 0
the message will be disabled. The system will announce
to the caller that the call will be disconnected. The event
will happen at the specified number of seconds before the
actual call disconnect takes place.
is below the specified value the system will remind the
caller with message.
To Edit IVR Profile
1 From the Navigation tree, select the IVR Management > IVR Profiles node.
2 From the right pane of the Navigator, select the IVR Profile to be edited.
3 From the Edit Menu select ‘Edit IVR Profile’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to enforce the modifications.
To Delete IVR Profile
1 From the Navigation tree, select the IVR Management > IVR Profiles node.
2 From the right pane of the Navigator, select the IVR Profile to be deleted.
3 From the Edit Menu, select the Delete menu item.
IVR Objects
Overview
IVR objects serve to build custom menus for the PBX system. They are attached to IVR
profiles. IVR objects consist of user selection filter, pointers to audio files to be played, and
action to be taken over the call (dialing, a number, dropping the call, connecting to other
menu etc). They allow for building multilevel menus. Each menu object should be
constructed from files having the same prefix and delimited with “.” e.g. menu.1, menu.2 ...
menu.N.
SysMaster Corporation
Page 78
72 IVR Objects
Parameters:
Path Name Defines the name of the menu.
Selection Specifies the digit used to select the menu entry.
Anouncement Audio File Name Specifies the name of the audio file to use. This name
together with the language abbreviation and the audio
format specifies the name of the file in use.
Selection Audio File Name Specifies whether the Intelligent Proxy option would be
activated or not. The Intelligent Proxy allows the VoIP
Gateway to relay audio frames between H323 and SIP
channels without using codec operations (encode/
decode).
Action Specifies the action that will be performed if the selection
is matched. There are several actions that can take place:
Dial
None
Drop Call
Directory Dial
Enter Path
Exit Path
Advanced Rule Specifies the Advanced Rule to be applied to the Call
Managing IVR Objects
To Add IVR Object.
1 From the Navigation tree, select the IVR Management > IVR Objects node.
2 From the Edit Menu select ‘Add IVR Object’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce the modifications.
To Edit IVR Object.
1 From the Navigation tree, select the IVR Management > IVR Objects node.
2 From the right pane of the Navigator, select the IVR Object to be edited.
3 From the Edit Menu select ‘Edit IVR Object’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to enforce the modifications.
To Delete IVR Object.
Filter. Advanced Rules allow for flexible enforcement of
the call filter based on time or endpoint availability.
1 From the Navigation tree, select the IVR Management > IVR Objects node.
2 From the right pane of the Navigator, select the IVR Object to be deleted.
SysMaster Corporation
Page 79
Chapter 8 IVR Setup 73
3 From the Edit Menu, select the Delete menu item.
Language Servers
Overview
Language Servers are used to store audio files through which the IVR logic will assemble
audio messages, e.g. announcing numbers, text messages and prompts. To this end, a
Language Server is pointed to in the IVR Profile. A Language Server actually represents a
directory on a remote server (TFTP protocol server) where audio files are stored. The
SysMaster gateway can also store locally multiple audio files. Each audio file has a name
defined by a short designation of the language as well as a part determining the type of
message it holds.
Each Language Server is identified by name. Remote Servers additionally designate remote
hosts with IP addresses and directories from where audio files are downloaded only one
time upon administrator’s request. Thus, two remote language servers can even use one and
same directory name, but the difference comes from the file name prefix of the files
determining the language of the files. This feature is especially helpful for TFTP servers that
can have only one publicly available directory.
On the other hand, Local Language servers do not use IP addressees, directory or Timeouts.
Their audio files are uploaded in one directory and are installed (i.e. activated) only when
required.
The download of the audio files from remote servers is (in different formats) performed by a
specified protocol (TFTP or HTTP). The actual download procedure is done manually by
the administrator of the gateway. Using the Sysmaster Console Navigator, administrators
should go to Language Servers > Language Servers > [Language Server item ] and from
the menu select the “Download Language Server“ menu command.
Depending on the Language Server type (Remote/Local), the following operations can be
performed:
File Name change (all type of Language servers).
Change of the Audio File contents
Each file can be uploaded and overwritten as well as played back.
Adding new files
This allows for adding more files outside of the objurgatory files on which the IVR
functionality relies.
Each audio file has its status. For Remote Language Servers it can be:
New - the file has never been downloaded;
Failed - the system has decided that the file has not been downloaded correctly or
that the download has failed entirely;
Downloaded - the file has been successfully downloaded.
For Local Language Servers it can be:
SysMaster Corporation
Page 80
74 L a n g u a g e S e r v e r s
New - the file has never been uploaded;
Not uploaded - the file has not been successfully uploaded;
Uploaded - the file has been successfully uploaded.
The purpose of all operations on both Local and Remote Language servers is for the IVR
logic to find installed files of one Language Server in one directory.
Parameters:
Name Defines the name of the language server that stores the
audio files.
Protocol Specifies the protocol to be used by the Language Server
Available protocols are:
TFTP
HTTP
IP Address Assigns an IP address to the Language Server
Directory Designates the directory within the Language Server
where audio files would be stored. The directory should
be named after the language abbreviation of the audio
files being stores in it. (e.g. ‘en’ for English, ‘fr’ for French,
‘sp’ for Spanish, etc) located in.
Timeout (1-60) Defines the period of time (in seconds) for timeout of the
server query. Language server timeout could range from
1 to 60 seconds.
Language Specifies the index (language abbreviation) of the audio
files located in the language server. When naming audio
files the following convention should be used:
<index_file.au>
, where index is the language abbreviation for the audio
file.
Foe example, the file en_welcome.au would store the
customized welcome message in English.
File Format Specifies the format of the IVR files stored in the
Language Server.
The selections available are:
PCM - uLaw 8kHz (.au)
GSM (.gsm)
SysMaster Corporation
Page 81
Chapter 8 IVR Setup 75
Language Type Defines the type of the Language Server.
Available options are:
Regular - file "md_and" is not used and numbers
are said like in English.
All Join - file "md_and" is played between every
major digits (20,30,...100,...1000) and "one" in one
hundred and one thousand is skipped.
Last Join - file "md_and" is played after last major
digit and "one" in one hundred and one thousand
is skipped.
Description Allows a short description (up to 71 characters) of the
conference profile to be entered.
Managing Language Servers
To Add Language Server.
1 From the Navigation tree, select the Language Server > Language Servers node.
2 From the Edit Menu select ‘Add Language Server’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce modifications.
To Edit Language Server.
1 From the Navigation tree, select the Language Server > Language Servers node.
2 From the right pane of the Navigator, select the Language Server to be edited.
3 From the Edit Menu select ‘Edit Language Server’.
A dialog box shows up.
4 Fill in the data and click on the Apply button to enforce modifications.
To Delete Language Server.
1 From the Navigation tree, select the Language Server > Language Servers node.
2 From the right pane of the Navigator, select the Language Server to be deleted.
3 From the Edit Menu, select the Delete menu item.
To Download Files of a Language Server.
1 From the Navigation tree, select the Language Server > Language Servers node.
2 From the right pane of the Navigator, select the Language Server whose files will be
downloaded.
3 From the Navigator menu, select the Download Language Server.
SysMaster Corporation
Page 82
76 Language Server Files
Language Server Files
Overview
Each Language Server file contains a piece of audio message. The IVR functionality of the
gateway uses these audio files to assemble longer messages such as numbers, text messages,
prompts etc. Language Server Files are organised in Language Servers that can be either
Local or Remote.
The format of the files should be one the following:
PCM - uLaw 8kHz - use “.au” extension;
GSM - use “.gsm” extension.
For more information, please refer to the Language Server section.
Parameters:
File Name Specifies the name of the audio file to be uploaded. The
name would be used as part of the file name for the audio
file.
Description Specifies a Second Language Server only used if multi-
Managing Language Server Files
To Add Language Server File.
1 From the Navigation tree, select the Language Server > Language Server Files >
[Language Server] node.
2 From the Edit Menu select ‘Add Language Server’.
A dialog box shows up.
3 Fill in the data and click on the Apply button to enforce modifications.
To Edit Language Server File.
1 From the Navigation tree, select the Language Server > Language Server Files >
[Language Server] node.
2 From the right pane of the Navigator, select the Language Server File to be edited.
3 From the Edit Menu select ‘Edit Language Server’.
A dialog box shows up.
For example, entering a name of 'welcome' would result
in a file name 'en_welcome.au'.
language IVR is desired.
4 Fill in the data and click on the Apply button to enforce modifications.
SysMaster Corporation
Page 83
Chapter 8 IVR Setup 77
To Delete Language Server File.
1 From the Navigation tree, select the Language Server > Language Server Files >
[Language Server] node.
2 From the right pane of the Navigator, select the Language Server File to be deleted.
3 From the Edit Menu, select the Delete menu item.
To Listen To a Language Server File.
1 From the Navigation tree, select the Language Server > Language Server Files >
[Language Server] node.
2 From the right pane of the Navigator, select the Language Server File to listen.
3 From the Edit Menu, select the Listen To Downloaded File menu item.
4 A dialog will popup to download the file. Click on the Save button.
5 Locate the saved file and play it in your favorite audio player supporting “.wav” files.
SysMaster Corporation
Page 84
78 Language Server Files
SysMaster Corporation
Page 85
Chapter 9 PBX Server 79
Chapter 9 PBX Server
Overview .............................................................................................................................................79
Call Filter Configuration ....................................................................................................................79
Inbound Profile Configuration ..........................................................................................................80
PBX Group Configuration..................................................................................................................82
Call Parking.........................................................................................................................................83
IVR Profile Configuration...................................................................................................................83
Default Route Configuration .............................................................................................................83
Managing PBX Groups ......................................................................................................................84
PBX Extensions..................................................................................................................................84
Managing PBX Extensions ................................................................................................................86
Overview
The document describes key provisioning issues related to the configuration of the
SysMaster PBX Server under the SM 7000 Platform.
SysMaster PBX server configuration involves:
Call Filter Configuration
Inbound Profile Configuration
PBX Group Configuration
The components above also depict the basic call flow for processing of calls by the
SysMaster PBX server.
The following sections provide basic explanation and interconnectivity guidelines for each
of the configuration tasks above.
Special importance will be given to PBX Groups and PBX Extensions configuration. For
more detail explanation on the rest, please refer to the corresponding chapters of the user
manual.
Call Filter Configuration
Call Filters could be viewed as firewalls granting or denying access to incoming calls. When
an incoming call is accepted the call will be associated with an inbound profile.
In turn, the inbound profile will specify a set of parameters according which the PBX server
will handle incoming calls.
Call Filters are the first processing component of the PBX Call Flow but the last to be
configured within the actual PBX work flow.
SysMaster Corporation
Page 86
80 Inbound Profile Configuration
A sample PBX Call Filter configuration could be:
Upon Call filter configuration special attention should be played to the selection of the
following components:
Endpoint
Inbound Profile
For more detail information on how to set up call filter, please refer to Chapter 3 “Call
Processing” of the user manual.
Inbound Profile Configuration
For the purpose of this document, an Inbound Profile specifies how the PBX incoming call
will be handled once accepted by the call filter.
Important parameters that could be designated within an Inbound Profile are:
IVR Profile Type
Routing Table Type
PBX Group Type
The IVR Profile Selected will contain the proper IVR logic that will be played on the side of
the PBX. Also an IVR Profile of type PBX is the place where the enhanced Auto Attendant
logic should be input.
The SysMaster Auto Attendant allows easy definition of complex PBX menus and submenus to be achieved.
Keeping within the tone of this section, next we will describe how to create IVR Profiles for
servicing the SysMaster PBX Server. In addition, IVR Objects will also be discussed.
An IVR Profile is associated with an incoming call through the Inbound Profile attached to
SysMaster Corporation
Page 87
Chapter 9 PBX Server 81
the designated Call Filter.
Each IVR Profile created is characterized by its type. Thus, the first step in configuring an
IVR Profile is selecting an appropriate IVR type. In our case, the IVR Profile will be of type
PBX.
A sample PBX type IVR Profile configuration is depicted below:
The value of the last field (Auto Attendant Starting Path) selected establishes the connection
between an IVR Profile and an IVR Object.
IVR Objects are used for tuning up the Auto Attendant feature of the PBX Server. In other
words, IVR Objects are used for building customized menus and sub-menus for the
SysMaster PBX Server. They are attached to the IVR Profile through the Auto Attendant
Starting Path. Visually, an Auto Attendant Starting Path could be conceived as a container
in which you can arrange different menus and submenus.
For example, we wish all menus or submenus to be within a single container called main.
Each menu and submenu will correspondent to a single IVR Object and be responsible for
the announcement of separate greetings. (i.e. Welcome to the PBX System, For sales press 1,
For support press 2, etc)
SysMaster Corporation
Page 88
82 PBX Group Configuration
The way we can put all menus and submenus within the single main container is by giving
them appropriate names. That is, the fist part of the name should consist of the name of the
container (in our case main) and the second one, preceded by a dot ì.î , should be the
arbitrary name given to the menu/sub-menu. (e.g. main.1; main.2; main.N1; main.N2)
In addition, an IVR Object or sub-menu could contain two types of prompts: non-action or
action prompts. A non-action prompt is comprised only of a single Announcement Audio
File that should be specified upon IVR Object configuration.
An action prompt on the other side contains an Announcement Audio File, a selection audio
file and a type of action attached to it.
The following IVR Object actions are available:
Dial
Drop Call
Directory Dial
Enter Path
Exit Path
Repeat Menu
Dial To Conference
Dial To Follow Me
Dial To ACD
IVR Function
Switch To
The action selected will be triggered each time the IVR Object is played and after a user has
made a selection.
An action prompt IVR Object is played in the following manner:
1 Announcement Audio File is played
2 User makes a selection
PBX Group Configuration
A PBX Group represents a blueprint outlining the basic behavior of all PBX extensions
contained within it and at the same time allows configuration of global PBX parameters
A PBX Group is identified by the following set of parameters:
Name Specifies the name of the PBX Group. The name could
contain any combination of alphabetical and numerical
characters.
Status Indicates whether the PBX Group created is enabled or
disabled. When disabled status is selected the PBX Group
would not function even though it has been defined.
SysMaster Corporation
Page 89
Chapter 9 PBX Server 83
First Park Number Allows the user to “park” a call at the specified number,
and access (dial) the “park” number through another
phone on the system to retrieve the call.
Last Park Number Specifies the last park number needed for establishing a
range within which the feature would operate. The
starting point of the range is the specified First Park
Number.
Parking Timeout Specifies the maximum time (in seconds) for which a call
can operate in the “park” mode.
IVR Profile Specifies the default IVR Profile that will be used by the
PBX Group.
Default Route Specifies the Default Routing table that will be used by
the PBX Group
There are several types of actions that could be configured within a PBX Group:
Call Parking
IVR Profile Connectivity
Default Route Connectivity
Call Parking
SysMaster 7000 GW provides a flexible way for conducting call park of calls.
Call park allows you to place a call on hold, by specifying a call park extension range, and
later retrieve it from another phone. For example, if a user dials a phone number that is
currently active, the receiving end can ìparkî the call to a call parking extension (i.e. 122) and
then the parked call could be retrieved by another phone on the system by dialing the call
parking extension specified.
To achieve such a scenario, you need to specify a range of call park numbers that will be
used as extensions. A call park range is defined by the first park number and last park
number specified.
IVR Profile Configuration
A PBX Group should be associated with an IVR Profile in order proper IVR logic to be
played. The IVR Profile selected should be the same as the one created under IVR Profiler
configuration.
IVR Profiles are attached to a PBX Group by simply selecting the proper profile from the list
of available ones.
Default Route Configuration
Default Route Configuration joins a PBX Group to an existing route on the system, thus
ensuring proper routing of calls. Routes are configured through the custom route window
under Call Routing > Routes > Add Custom Route.
SysMaster Corporation
Page 90
84 PBX Extensions
For detailed information on route configuration refer to Chapter 6 “Call Routing” of the
user’s manual.
Managing PBX Groups
To Add PBX Group
1 From the Navigation tree, select the PBX > PBX Groups node.
2 From the Edit Menu select ‘Add PBX Group’
A dialog box shows up.
3 Fill in the data, and click on the Apply button to enforce the modifications.
To Edit PBX Group
1 From the Navigation tree, select the PBX > PBX Groups node.
2 From the right pane of the Navigator, select the PBX Group to be edited.
3 From the Edit Menu select ‘Edit PBX Group’.
A dialog box shows up.
4 Fill in the data, and click on the Apply button to enforce modifications.
To Delete PBX Group
1 From the Navigation tree, select the PBX > PBX Groups node.
2 From the right pane of the Navigator, select the PBX Group to be deleted.
3 From the Edit Menu, select the Delete menu item.
PBX Extensions
Once a PBX Group has been created, PBX Extensions could be added to it. All PBX
Extensions will inherit the default behavior of the group they belong to.
A PBX Extension is associated with a single party within a PBX Group. One or more PBX
Extension could be contained within a group.
PBX Extension parameters that could be configured are:
Directory Name Dialing Initials Directory Name Dialing Initials allows in-bound callers
to have their calls routed to the appropriate person
without having to enter the person’s extension number.
Once implemented the feature will conduct a first and
last name directory look up and determine the
appropriate party to receive the call.
The only setting that needs be specified is the value of the
Directory Name Dialing Initials field. In it you can enter
the first or last name of the party that will participate in
the directory name look up.
SysMaster Corporation
Page 91
Chapter 9 PBX Server 85
Status The Status flag serves as an indicator of whether a PBX
Extension is operational or not. Selecting a Status
Enabled ensures that the PBX Extension is active, where
status disabled indicates that a PBX Extension has been
created but is still not functional.
Enable DND When checked DND (Do Not Disturb) will be enabled.
The functionality allows you to block incoming calls from
ringing the terminal. Once enabled the DND feature will
trigger either a call forward of the blocked incoming
number or a transfer of the incoming call to the specified
voicemail number below.
For more information refer to the Forward and Voicemail
Number field explanations.
Hide Caller ID When selected the Caller ID of the calling party will not
be displayed.
Enable Call Waiting When checked call waiting will be enabled.
Enable Call Screening The option activates Call Screening.
Enable Call Recording When selected the Caller ID of the calling party will not
be displayed.
Forward Call Forward is used in conjunction with the DND
functionality. Call Forward specifies a set of conditions
for which forwarding of calls will be enabled.
Available conditions are:
If Busy/No Answer
If Busy
If No Answer
Call Forward will be triggered once the selected
condition has been met.
Forced DND Rule The option allows you when the enabled DND Rule will
take effect.
The field allows you to enter a number to which calls will
be forwarded. Call forwarding will be enabled based on
the forward rule specified.
The option will allow you to enter a new number based
on the DND rule selected.
The field allows you to enter a number to which calls will
be forwarded. Call forwarding will be enabled based on
the forward rule specified. The field allows you to enter a
number to which calls will be forwarded. Call
forwarding will be enabled based on the forward
SysMaster Corporation
Page 92
86 PBX Extensions
Forward Number The option allows you to specify a forward number that
calls will be redirected to in case the PBX Extension is
unavailable.
Number forward is processed based on the Forward rule
selected above.
Voicemail Number The option allows you to specify a voicemail number that
calls will be redirected to.
Speed Dials Speed dial is used for faster dialing of numbers. The
feature allows administrators to create system level dial
codes that can be used by any telephone terminal
connected to the PBX. For enabling speed dial
capabilities two sets of numbers must be defined (e.g. 1 =
1221843553). The number to the left of the equal sign
denotes the speed-dial number that needs to be called.
The number to the right is the phone number to be
replaced by the speed dial number.
Timeout In the event that a PBX Extension is not reachable, the
system will periodically initiate a query trying to locate
the PBX Extension. The time the PBX Extension query
will be initiated for is controlled by the value of the
Timeout field. Once the timeout is reached, the system
will cease initiating the PBX Extension query.
Managing PBX Extensions
To Add PBX Extensions
1 From the Navigation tree, select the PBX > PBX Extensions > [PBX Extension] node.
2 From the Edit Menu select ‘Add PBX Extension’
A dialog box shows up.
3 Fill in the data, and click on the Apply button to enforce the modifications.
To Edit PBX Extensions
1 From the Navigation tree, select the PBX > PBX Extensions > [PBX Extension] node.
2 From the right pane of the Navigator, select the PBX Extension to be edited.
3 From the Edit Menu select ‘Edit PBX Extension’.
A dialog box shows up.
4 Fill in the data, and click on the Apply button to enforce modifications.
To Delete PBX Extensions
1 From the Navigation tree, select the PBX > PBX Extensions > [PBX Extension] node.
2 From the right pane of the Navigator, select the PBX Extension to be deleted.
3 From the Edit Menu, select the Delete menu item.
SysMaster Corporation
Page 93
Chapter 9 PBX Server 87
To Delete All PBX Extensions
1 From the Navigation tree, select the PBX > PBX Extensions node.
2 From the Edit Menu, select the Delete All menu item.
SysMaster Corporation
Page 94
88 PBX Extensions
SysMaster Corporation
Page 95
Chapter 10 Voice Mail Server 89
Chapter 10 Voice Mail Server
Introduction ........................................................................................................................................89
Voice Mail Groups ..............................................................................................................................90
Voice Mail Boxes ................................................................................................................................91
Introduction
Overview
GW 7000 supports H323, SIP and PSTN protocols to allow flexible call processing. Callers
can connect to the Voice Mail platform via IP or PSTN or Web and have full access to all
advanced Voice Mail functionality. International and long-distance VoIP callers can also
leave, retrieve, and manage messages in real-time from a remote Voice Mail server. In
addition, the platform allows telecommuters and corporate field associates to take
advantage of Voice Mail functionality while traveling or working from remote locations.
GW 7000 unique features flexible price structure and high return on investment.
Features
Unlimited Number of Mailboxes
Long-distance and International Voice Mail Support
H323,SIP and PSTN Protocol Support
Complete Voice Mail Functionality over IP and PSTN
Password/PIN Based security
Web and Phone Management Access for Mailbox Users and Administrators
Voice Mail Email Exchange and Notification
Voice Mail Forwarding
RADIUS Billing Interface
Managed Services and Virtual Platform Partitioning Support
Unlimited Number of Mail Boxes
GW 7000 supports an unlimited number of mailboxes with pre-set size quotas. The
mailboxes are grouped for flexible administrative templates and easy management mailbox
users can also be related to PBX accounts to allow single access to PBX and Voice Mail
functionality.
Long-Distance and International Voice Mail
GW 7000 allows long-distance and international users to support virtual mailboxes outside
the country of their residence. In fact users can be anywhere in the world and retrieve their
messages via Web, Email, or VoIP. This allows extremely flexible platform implementation
and dynamic virtual office support.
SysMaster Corporation
Page 96
90 Voi ce Mail Groups
H323, SIP, PSTN/TDM Web, Email Access Support
GW 7000 allows callers to leave a message for unavailable users once they connect to the
platform via VoIP or PSTN. In addition, users can manage their greetings or passwords via
VoIP or PSTN as well. The messages can be retrieved via Web, VoIP, PSTN, and Email to
allow flexible mailbox management and low retrieval cost.
Web and Phone Mailbox Management
GW 7000 allows flexible mailbox management for all system users to provide them with the
necessary tools to modify greetings and passwords, to delete and forward messages, and to
setup mailbox options. Administrators can expire, lock, and restrict user accounts to secure
the system integrity.
RADIUS Billing Interface
GW 7000 supports Radius Billing Interface to SysMaster VM2000 Billing Platform. This
allows the Voice Mail server to support real time billing procedures where as all inbound
voice mail messages are accounted and billed for.
Voice Mail Groups
Parameters:
Group Name Specifies the name of the voice mail group to be added.
Group Status: Enable / Disable Verifies the password.
Max Messages Specifies the maximal number of voice messages
Max Message Time Specifies the maximal duration of the message in
Lock Attempts Allocates a number for the lock attempts to be performed.
Audio Format Specifies the audio format of the voice mail file.
Quick Menu Specifies whether a quick menu will be included in the
Enable Administration If checked, remote administration could be performed.
Authentication Method Specifies how the voice mail group would be
contained within the group.
seconds.
created voice mail group.
authenticated.
RADIUS Group Specifies the Radius Server Group of the Radius Server to
Description Allows a short description (up to 71 characters) of the
use
conference profile to be entered
SysMaster Corporation
Page 97
Chapter 10 Voice Mail Server 91
Managing Voice Mail Groups
To Add Voice Mail Group
1 From the Navigation tree, select the Voice Mail > Voice Mail Groups node.
2 From the Edit Menu select ‘Add Voice Mail Group’
A dialog box shows up.
3 Fill in the data, and click on the Apply button to enforce the modifications.
To Edit Voice Mail Group
1 From the Navigation tree, select the Voice Mail > Voice Mail Groups node.
2 From the right pane of the Navigator, select the Voice Mail Group to be edited.
3 From the Edit Menu select ‘Edit Voice Mail Group’.
A dialog box shows up.
4 Fill in the data, and click on the Apply button to enforce modifications.
To Delete Voice Mail Group
1 From the Navigation tree, select the Voice Mail > Voice Mail Groups node.
2 From the right pane of the Navigator, select the Voice Mail Group to be deleted.
3 From the Edit Menu, select the Delete menu item.
Voice Mail Boxes
Parameters:
Box Name Specifies the name of the voice mail box to be configured.
Extension Number Denotes the extension number to be used for accessing
Email Specifies the email of the administrator administering the
Attach Message to Email Specifies the maximal duration of the message in
Client Name Identifies the user name to be used upon administering
Password Assigns a password necessary for accessing the Voice
the Mail Box.
Mail Box.
seconds.
the Mail Box.
Mail Box.
Confirm Password Confirms the password entered above.
Password Expiration Specifies the expiration period for the Voice Mail Box
password. After the end of the expiration period the
assigned password will no longer be valid and a new one
must be specified.
SysMaster Corporation
Page 98
92 Voice Mail Boxes
Mail Box Expiration (days) Specifies the expiration period for the Voice Mail Box.
After the end of the expiration period the Voice Mail Box
will no longer be active.
Box Status Enables or disables the current Voice Mail Box.
Web Management Enables Web Management.
Email Management Enables Email Management. When enabled Email
Management would allow messages to be attached to
emails sent to the address specified in the email field.
Custom Announcement Enables custom announcements.
Locked When checked the Voice Mail Box would be locked.
Disable "Delete" When checked users will not be able to manually delete
messages contained within a Voice Mail Box.
Disable "Delete All" When checked users will not be able to manually delete
all messages contained within a Voice Mail Box.
Disable "Delete Forwarded" When checked users will not be able to manually delete
forwarded messages contained within a Mail Box.
Voice Mail Group Associated the Voice Mail Box with an already created
Managing Voice Mail Boxes
To Add Voice Mail Box
1 From the Navigation tree, select the Voice Mail > Voice Mail Boxes node.
2 From the Edit Menu select ‘Add Voice Mail Box’
A dialog box shows up.
3 Fill in the data, and click on the Apply button to enforce the modifications.
To Edit Voice Mail Box
1 From the Navigation tree, select the Voice Mail > Voice Mail Boxes node.
2 From the right pane of the Navigator, select the Voice Mail Box to be edited.
3 From the Edit Menu select ‘Edit Voice Mail Box’.
A dialog box shows up.
4 Fill in the data, and click on the Apply button to enforce modifications.
To Delete Voice Mail Box
Voice Mail Group.
1 From the Navigation tree, select the Voice Mail > Voice Mail Boxes node.
2 From the right pane of the Navigator, select the Voice Mail Box to be deleted.
3 From the Edit Menu, select the Delete menu item.
SysMaster Corporation
Page 99
Chapter 11 Callback Server 93
Chapter 11 Callback Server
Overview .............................................................................................................................................93
CallBack Server Setup .......................................................................................................................99
Overview
The SysMaster gateway can act as a callback server. A callback server operates the following
way:
1 A user sends to the gateway a PIN, a source number (the number from which the user
wants to place a call) and destination number through a web request, email, or SMS
message.
2 Next the call back server authenticates the user, calls back to the source number that the
user has provided and at the same time calls the destination number. This way the
gateway connects both parties. Alternatively, the destination number can be provided
later after the gateway calls back the user first. Then the user will hear an IVR message
prompting him/her to enter the destination number.
The callback server can receive requests from:
Web based HTTP request
SMS messages
Email messages
PSTN calls (ANI authentication/DID)
There are two ways of authentication:
PIN based authentication (web callback, email callback, sms callback)
ANI (Caller ID, source number)/DNIS based authentication (PSTN callback)
SysMaster Corporation
Page 100
94 Overview
General Callback Server Processing
The general operation of the callback server includes the following steps:
1 The gateway receives PIN, prefix, source number and destination number through the
web interface or SMS or email. The prefix and destination number are optional. If the
destination number is omitted, it should be later provided.
2 (Step 1) Upon receiving the callback request, the gateway internally creates an incoming
call leg with the following parameters:
ANI[web] - source number.
DNIS[web] - destination number
According to the values above:
ANI[web] = 12315101234567
DNIS[web] = 22201234567
3 The gateway intercepts the internally created incoming call with a Callback filter that
matches against the callback server endpoint. The primary Call Filter must have higher
priority than the secondary Callback Filter (Callback filters should be created in pairs)
so that it takes call processing control. The primary Call Back filter should redirect to an
already created Inbound Profile.
SysMaster Corporation
Loading...