Sipelia User Guide - Trinity Fire &

Sipelia User Guide
2.0 GA
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Copyright notice
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The contents of this guide are furnished for informational use only and are subject to change without
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"Genetec", "Omnicast", "Synergis", "Synergis Master Controller", "AutoVu", "Federation", "Stratocast",
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Document information
Document title: Sipelia User Guide
Document number: EN.704.004-V2.0B(3)
Document update date: February 26, 2015
You can send your comments, corrections, and suggestions about this guide to
documentation@genetec.com.
About this guide
This guide is intended for Sipelia administrators. It describes how to set up, configure, and manage the
Sipelia module as part of your Security Center system.
Notes and notices
The following notes and notices might appear in this guide:
• Tip. Suggests how to apply the information in a topic or step.
• Note. Explains a special case, or expands on an important point.
• Important. Points out critical information concerning a topic or step.
• Caution. Indicates that an action or step can cause loss of data, security problems, or performance
issues.
• Warning. Indicates that an action or step can result in physical harm, or cause damage to hardware.
IMPORTANT: Topics appearing in this guide that reference information found on third-party websites
were accurate at the time of publication, however, this information is subject to change without prior
notice to Genetec.
Contents
Preface: Preface
Copyright notice .
About this guide .
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Chapter 1: Getting started
What is Sipelia? . . . . . . . . . . . . . . . . . . . . . .
How licensing works in Sipelia . . . . . . . . . . . . . . . . . .
Deploying Sipelia . . . . . . . . . . . . . . . . . . . . .
How to use Sipelia in Security Desk . . . . . . . . . . . . . . . . .
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Chapter 2: Installation
About Sipelia Server . . .
Default ports for Sipelia Server
Sipelia Client . . . . .
Default ports for Sipelia Client
Installing Sipelia Server . .
Installing Sipelia Client . .
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Creating the Sipelia Plugin role . . . . . . . . . . . . . . . . .
Configuring the system communication service . . . . . . . . . . . . . .
Configuring the SIP port of Sipelia Server . . . . . . . . . . . . . . .
Defining the ranges of SIP phone extensions . . . . . . . . . . . . . .
Recording the audio and video of call sessions . . . . . . . . . . . . . .
Configuring SIP accounts for Security Center users . . . . . . . . . . . . .
Allowing users to see pictures of other users . . . . . . . . . . . . . .
Associating Security Center cameras with users . . . . . . . . . . . . .
Adding SIP intercoms . . . . . . . . . . . . . . . . . . . .
Associating Security Center entities with SIP intercoms . . . . . . . . . . . .
Registering your SIP intercom with Sipelia Server . . . . . . . . . . . . .
Ring groups . . . . . . . . . . . . . . . . . . . . . .
Creating basic ring groups . . . . . . . . . . . . . . . . . . .
Creating custom ring groups . . . . . . . . . . . . . . . . . .
Configuring devices for voice and video calls . . . . . . . . . . . . . .
Configuring two-way communication between Sipelia Server and other SIP servers . . . . .
Configuring your SIP intercom to call a specific extension . . . . . . . . . . .
Adding SIP intercom icons to a Plan Manager map . . . . . . . . . . . . .
Configuring a network interface with the highest priority . . . . . . . . . . .
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Chapter 3: Configuration
Chapter 4: SIP trunks and dial plans
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Adding SIP trunks . . . . . . . . . . . . . . . . . . . . .
Dial plans . . . . . . . . . . . . . . . . . . . . . . .
Regular expressions in Sipelia . . . . . . . . . . . . . . . . . .
Defining dial plan rules . . . . . . . . . . . . . . . . . . .
Importing dial plans . . . . . . . . . . . . . . . . . . . .
Dial plan scenario 1: Forwarding to a SIP trunk all calls starting with a prefix . . . . . .
Dial plan scenario 2: Reserving a range of SIP extensions for local calls . . . . . . . .
Dial plan scenario 3: Reserving a range of SIP extensions for calls to a SIP trunk . . . . . .
Dial plan scenario 4: Replacing source SIP extensions . . . . . . . . . . . .
Dial plan scenario 5: Removing prefix on source SIP extensions from a SIP trunk . . . . .
Dial plan scenario 6: Forwarding calls to another SIP extension on schedule . . . . . . .
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Chapter 5: Troubleshooting
Troubleshooting: Unable to establish communication with server . . . . . . . . .
Troubleshooting: Message broker connection failed . . . . . . . . . . . .
Troubleshooting: Cannot add SIP intercom devices . . . . . . . . . . . . .
Troubleshooting: Cannot see the Sipelia icon in the notification tray . . . . . . . .
Troubleshooting: Security Desk cannot connect to Sipelia Server . . . . . . . . .
Troubleshooting: Cannot register to Sipelia Server from Security Desk . . . . . . . .
Troubleshooting: Cannot make calls between two SIP endpoints . . . . . . . . .
Troubleshooting: No video displayed during calls . . . . . . . . . . . . .
Troubleshooting: Audio and video not being recorded . . . . . . . . . . . .
Troubleshooting: Users cannot view recorded video . . . . . . . . . . . .
Additional resources
Appendix A: Common VoIP terms .
Common VoIP terms .
Glossary .
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Where to find product information . . . . . . . . . . . . . . . .
Technical support . . . . . . . . . . . . . . . . . . . . .
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v
1
Getting started
This section includes the following topics:
•
"What is Sipelia?" on page 2
•
"How licensing works in Sipelia" on page 4
•
"Deploying Sipelia " on page 5
•
"How to use Sipelia in Security Desk" on page 6
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1
Getting started
What is Sipelia?
Sipelia is a core module of Security Center that allows Security Center users to make, receive, and
manage SIP-based voice and video calls over a network. Running on the open source Session Initiation
Protocol (SIP), Sipelia also integrates existing video and access control platforms with intercom
systems, and allows users to log call activities.
Main features
With Sipelia installed within Security Center, you can do the following:
• Connect standard USB headsets and webcams to Security Desk workstations, so that you can make
voice and video calls through Security Center.
• Receive incoming call notifications directly through the notification tray in Security Desk.
• Initiate, answer, forward, place on hold, or cancel calls from a dedicated call dialog box.
• Generate reports to investigate the activities within specific call sessions.
• Watch call sessions that have associated video.
• Control cameras, doors, zones, and device outputs during a call.
• Deploy a SIP-based solution that makes it easy to leverage your existing communications
infrastructure.
• Connect to SIP intercom devices, intercom exchange servers, and mobile apps through the SIP
standard.
• Create a customizable list of contacts, so that users can quickly call their contacts. Contact lists can
include other Security Center users, as well as SIP devices.
• Create ring groups so that multiple Security Center users and SIP entities can receive incoming calls
at the same time or one after another until a user takes the call.
Typical applications
A Sipelia integration within Security Center can help you in the following applications:
• Responding to and investigating an emergency
• Responding to employees who have lost their cards
• Granting access to high-security rooms
• Monitoring and managing who enters and exits a parking lot
• User to user video calls, making communication more efficient
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2
Getting started
Overview of Sipelia Communications Management
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3
Getting started
How licensing works in Sipelia
Sipelia requires a set of licenses that allow the installation and use of the plugin in Security Center and
to enable more advanced features.
Sipelia requires the following licenses:
• GSC-Sipelia-Base: The base system license is required to install the Sipelia plugin on Security Center
and to allow the operators to make and receive calls.
• GSC-Sipelia-1SIP-STD: The standard license allows a SIP device (such as a SIP intercom) to be used
in Security Center, whether it registers directly to Sipelia Server or is made available via a SIP trunk.
You need one standard license per SIP device that you will be adding to your system.
• GSC-Sipelia-1SIP-ADV: The advanced license enables the recording of call sessions and failover of
the Sipelia Plugin role to a second server when needed. You need one advanced license for each
standard license that you will be adding to your system.
NOTE: If the standard and advanced license counts do not match, the system will choose the lower
count for both types of licenses, as shown in the example below.
• GSC-Sipelia-1Trunk: The trunk license allows you to add and configure a SIP trunk. You need one
trunk license for each SIP trunk that you will be adding to your system.
Example
The following sample license shows how the system will apply the license counts when they do not
match:
License type
Number of licenses installed
Number of licenses applied
GSC-Sipelia-Base
1
1
GSC-Sipelia-1SIP-STD
100
50
GSC-Sipelia-1SIP-ADV
50
50
GSC-Sipelia-1Trunk
4
4
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4
Getting started
Deploying Sipelia
To integrate SIP-based communication into Security Center, so that users can communicate through
VoIP, you can deploy Sipelia as part of your Security Center system.
Before you begin
• Read the Sipelia 2.0 GA Release Notes.
• Familiarize yourself with the default ports for Sipelia Server.
• Familiarize yourself with the common VoIP terms and the Sipelia glossary of terms that are used
throughout this guide.
To deploy Sipelia:
1 Install Sipelia Server .
2 Configure the system communication service.
3 Configure the SIP port of Sipelia Server .
4 Define the ranges of your SIP phone extensions.
5 Configure the audio and video recording of call sessions.
6 Configure SIP accounts for your Security Center users.
7 Create basic ring groups.
8 Create custom ring groups.
9 Add your SIP intercoms.
10 Register your SIP intercoms with Sipelia Server.
11 Install Sipelia Client on each of the Security Desk workstations that will run Sipelia.
12 Configure your devices for voice and video calls in Security Desk.
13 Configure two-way communication between Sipelia Server and other SIP servers.
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5
Getting started
How to use Sipelia in Security Desk
In Security Desk you can make and receive calls, manage your list of contacts and generate reports
about previous calls.
To learn more on how to use Sipelia in Security Desk, you can view the following videos:
• Sipelia user interface overview.
• Sipelia call management.
• Sipelia call report task overview.
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2
Installation
This section includes the following topics:
•
"About Sipelia Server" on page 8
•
"Default ports for Sipelia Server " on page 9
•
"Sipelia Client" on page 11
•
"Default ports for Sipelia Client " on page 12
•
"Installing Sipelia Server " on page 13
•
"Installing Sipelia Client" on page 15
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Installation
About Sipelia Server
Sipelia Server is the SIP server component of Sipelia. It receives and administers information about the
different SIP endpoints, and essentially facilitates the communication between two or more endpoints
that are communicating in a SIP environment. Sipelia Server also collates and stores important data,
such as contact list information, SIP server settings, and call session recordings.
Sipelia Server is a server plugin ( ) that must be run by a Security Center Plugin role. As a result,
Sipelia Server must be installed on every Security Center server where you intend to host this Plugin
role.
Sipelia Server stores the following data:
• User options
• List of contacts
• Session information (for example: users, time and duration of phone calls, and associated entities)
• Audio and video files related to call sessions
• Phone extension configurations for users and devices
• SIP server settings
• Ring group configuration
• SIP trunk configuration
• Dial plan configuration
• Recording settings for users and devices
• Security Center events that are linked to the SIP intercom entity
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Installation
Default ports for Sipelia Server
To ensure that Sipelia works properly, the ports used by Sipelia Server modules must be open and
redirected for firewall purposes.
IMPORTANT: When configuring ports, make sure that the ports are open and that they are not being
used by another application on the same workstation. For example, if Sipelia Server is installed on the
same machine that hosts the Genetec Server, you cannot use the same port that is already being used
by Security Center or another application.
Sipelia Server
component
Default
port
number
Protocol
Description
System
communication port
5672
TCP
The port that Sipelia uses to communicate with
the RabbitMQ system communication service. The
default value is 5672, which is a standard of the
RabbitMQ service configuration.
Configuration service
port
8201
TCP
The port that Config Tool uses to communicate
configuration settings with the Sipelia Server. The
default value is 8201. If there are issues with this
port number, you can enter another applicable
value.
Session transfer port
8202
TCP
The port that Sipelia Server uses to download
recordings of call sessions to the Call report task in
Security Desk. The default value is 8202. If there are
issues with this port number, you can enter another
applicable value.
Sipelia Server: SIP
port
5060
TCP (SIP)
The port used to enable the SIP protocol on
Sipelia Server. As a result, it is the basis of all SIP
communication in Sipelia. The default value is 5060.
Every SIP endpoint, such as softphones and SIP
intercoms, that needs to connect to the Sipelia
Server must have this port value in their respective
configurations.
SIP trunks: SIP port
5060
TCP (SIP)
The port used by the SIP trunk to communicate with
the Sipelia Server. Because SIP trunks are SIP servers,
the default value is 5060.
SIP trunks are needed if you have a device that is
connected to an external PBX, and you want to
connect this device to Sipelia. In this case, the PBX
will be in trunk mode with Sipelia Server.
Sipelia Server UDP
port range
20000-
UDP (SIP)
20500
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The port range for the User Diagram Protocol (UDP).
The UDP ports are used by the different SIP clients to
send and receive communication data. The default
9
Installation
Sipelia Server
component
Default
port
number
Protocol
Description
range is from 20000 to 20500. It is recommended to
keep the default settings, and to change them only if
Sipelia logs any port-related issues about making or
receiving calls with Security Desk.
The UDP port range used by Sipelia Server is set with
the MinimumPortRange and MaximumPortRange
properties found in C:\ProgramData\Genetec Sipelia
2.0\SipServer\SipServer.config.
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Installation
Sipelia Client
Sipelia Client is the softphone component of Sipelia. As a result, it installs the various user interface
features of the Sipelia module, such as the call dialog box and conversation window.
Sipelia Client installs the following:
• Notification tray
• Call dialog box
• Conversation window
• Call report task
Although not mandatory, it is recommended that you install Sipelia Client after installing and deploying
Sipelia Server. If Sipelia Client is installed before Sipelia Server, the user interface in Sipelia will not be
enabled.
Sipelia Client must be installed on every Security Desk workstation that is running Sipelia, thus turning
Security Desk into a SIP client (or softphone). The following image shows some of the components of
Sipelia Client.
Call report task
Call notification
Conversation window
Call dialog box
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Installation
Default ports for Sipelia Client
To ensure that Sipelia works properly, the ports used by Sipelia Client must be properly set in Security
Desk.
IMPORTANT: When configuring ports, make sure that the ports are open and that they are not being
used by another application on the same workstation.
Sipelia Client
component
Default
port
number
Protocol
Description
Advanced: UDP port
range
20000-
UDP (SIP)
The port range for the User Diagram Protocol (UDP).
The UDP ports are used by the different SIP clients to
send and receive communication data. The default
range is from 20000 to 20500. It is recommended to
keep the default settings, and to change them only if
Sipelia logs any port-related issues about making or
receiving calls with Security Desk.
20500
You can change the UDP port range by clicking
Options > Sipelia > Advanced in Security Desk.
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Installation
Installing Sipelia Server
To integrate SIP functions into Security Center, and allow your system to store data such as phone book
contacts and the recordings of call sessions, you must first install Sipelia Server before configuring the
Sipelia module in Config Tool.
Before you begin
Make sure of the following:
• Your servers meet the hardware requirements described in the Sipelia Release Notes.
• Config Tool is installed on the system on which you plan to install Sipelia Server.
IMPORTANT: It is recommended to install Sipelia Server on a dedicated Security Center expansion
server. Refer to the Security Center Administrator Guide for details on how to add an expansion server
to your Security Center system.
What you should know
Sipelia Server is a server plugin ( ) that must be run by a Security Center Plugin role. As a result,
Sipelia Server must be installed on every Security Center server where you intend to host this Plugin
role.
Although not mandatory, it is recommended that you install Sipelia Client after installing and deploying
Sipelia Server. If Sipelia Client is installed before Sipelia Server, the user interface in Sipelia will not be
enabled.
To install Sipelia Server:
1 Download the product from GTAP (https://gtap.genetec.com). You need a username and password
to log on to GTAP.
2 Double-click on setup.exe to run the product's setup.
The product's InstallShield Wizard dialog box opens.
3 Select the installation language, and then click OK.
This language selection does not limit the language availability of the installed software. The Sipelia
user interface appears in the language that is selected for Security Center.
4
5
6
7
8
Click Next.
Read the license agreement, accept the terms, and then click Next.
Select a folder location to install the product, and then click Next.
In the Custom Setup dialog box, select Server, and then click Next.
Click Install.
The installation might take a few minutes.
9 Once completed, select Restart Genetec Server, and then click Finish.
IMPORTANT: You must restart the Genetec Server for the system to detect that a new plugin has
been installed.
Restarting the Genetec Server causes a short interruption of the service on the server. If you cannot
afford to interrupt the service at this time, you can restart the Genetec Server later, as long as you
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Installation
do so before configuring Sipelia in Config Tool. To avoid interruptions, it is recommended to install
Sipelia Server on a dedicated Security Center expansion server.
Along with Sipelia Server, the RabbitMQ system communication service, which comes bundled with the
Sipelia Server installation, is installed automatically. RabbitMQ is the communication channel between
Security Desk and Sipelia Server. It is an essential service for ensuring that Sipelia works properly.
After you finish
In Config Tool, create the Sipelia Plugin role.
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Installation
Installing Sipelia Client
To turn Security Desk into a SIP client and use the various features of the Sipelia module, you must
install Sipelia Client on every Security Desk workstation that is running Sipelia.
Before you begin
Make sure of the following:
• Sipelia Server is installed on your Security Center system.
• Security Center Client is installed on the computer you want to install Sipelia Client.
• On computers that provide multiple network interfaces (cards), the network interface to be used by
Sipelia Client must be configured with the highest priority.
What you should know
Although not mandatory, it is recommended that you install Sipelia Client after installing and deploying
Sipelia Server. If Sipelia Client is installed before Sipelia Server, the user interface in Sipelia will not be
enabled.
To install Sipelia Client:
1 Download the product from GTAP (https://gtap.genetec.com). You need a username and password
to log on to GTAP.
2 Double-click on setup.exe to run the product's setup.
The product's InstallShield Wizard dialog box opens.
3 Select the installation language, and then click OK.
This language selection does not limit the language availability of the installed software. The Sipelia
user interface appears in the language that is selected for Security Center.
Click Next.
Read the license agreement, accept the terms, and then click Next.
Select a folder location to install the product, and then click Next.
In the Custom Setup dialog box, expand the Client node.
If you have Plan Manager installed and you want to add SIP intercoms on maps, select Plan
Manager Intercom Object, and then click Next.
9 Click Install.
4
5
6
7
8
The installation might take a few minutes.
10 Once completed, click Finish.
After you finish
Configure your devices for voice and video calls in Security Desk.
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3
Configuration
This section includes the following topics:
•
"Creating the Sipelia Plugin role" on page 17
•
"Configuring the system communication service" on page 18
•
"Configuring the SIP port of Sipelia Server " on page 19
•
"Defining the ranges of SIP phone extensions" on page 20
•
"Recording the audio and video of call sessions" on page 22
•
"Configuring SIP accounts for Security Center users" on page 23
•
"Allowing users to see pictures of other users" on page 25
•
"Associating Security Center cameras with users" on page 26
•
"Adding SIP intercoms" on page 28
•
"Associating Security Center entities with SIP intercoms" on page 30
•
"Registering your SIP intercom with Sipelia Server" on page 32
•
"Ring groups" on page 33
•
"Creating basic ring groups" on page 34
•
"Creating custom ring groups" on page 36
•
"Configuring devices for voice and video calls" on page 39
•
"Configuring two-way communication between Sipelia Server and other SIP servers" on page 40
•
"Configuring your SIP intercom to call a specific extension" on page 42
•
"Adding SIP intercom icons to a Plan Manager map" on page 43
•
"Configuring a network interface with the highest priority" on page 44
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Creating the Sipelia Plugin role
Once you have installed Sipelia Server on a Security Center server machine, you can create the Sipelia
plugin role in Config Tool.
Before you begin
Make sure that Sipelia Server is installed.
To create the Sipelia Plugin role:
1 Log on to Security Center with Config Tool.
2 Open the Plugins task, and then click Plugin ( )
The Creating a role: Plugin wizard appears.
3 In the Server drop-down list, select the server that is going to host the Sipelia Server role.
4 In the Select plugin type field, select Sipelia.
5 Enter the values for Database server and Database for the Sipelia database, or use the default
values which are already provided.
6 Click Next, enter the entity name and description, and then select the desired partition for this role.
7 Click Next and check that the information that you have entered is correct.
8 Click Create, and then click Close.
Sipelia appears in the list of Plugin roles ( ). It might take a few minutes for the role to create its
database.
After you finish
If deploying Sipelia, configure the system communication service.
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Configuring the system communication service
To ensure that the communication between Security Desk and Sipelia Server works properly, you must
configure the system communication service on the computer that hosts the Sipelia Server and make
sure that the proper IP address is used.
What you should know
The system communication service for Sipelia is RabbitMQ. RabbitMQ is an open source, external
Windows service that allows applications that are running on different servers, and at different times,
to communicate across dissimilar networks and computers, regardless of whether they are online.
Sipelia Server needs this service to properly communicate with Security Desk. As a result, this is an
essential requirement for a Sipelia installation.
The IP address of Sipelia Server must be the first private address listed in the Security Center server
properties.
To configure the system communication service:
1 Log on to Security Center with Config Tool, and then open the Plugins task.
2 Select the Sipelia Plugin role, and then click General.
3 Set the following:
• Communication service address: The hostname or IP address of the computer that hosts the
system communication service, RabbitMQ. The RabbitMQ service is an operating requirement
for the Sipelia Server, and it is installed automatically on the computer onto which Sipelia Server
is installed. The default value is localhost. Change the default value only if you want to point to
the RabbitMQ service that is running on another computer.
In case of any DNS issues with the default setting of localhost, you can enter the IP address of
the computer that hosts the RabbitMQ service.
• Communication service port: The port that Sipelia uses to communicate with the RabbitMQ
system communication service. The default value is 5672, which is a standard of the RabbitMQ
service configuration.
4 Open the Network task, and then select the server that is hosting the Sipelia Plugin role.
5 Click Properties, and make sure that the first IP address shown in Private addresses, that is, the IP
address listed at the top, is the one that you want to be used by the server.
The system communication service is configured. If there are any issues with the RabbitMQ connection,
descriptions of these issues appear in the Plugin Diagnose window.
For a description of the other ports shown on the General page, see Default ports for Sipelia Server on
page 9.
After you finish
If deploying Sipelia, configure the SIP port of Sipelia Server.
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Configuration
Configuring the SIP port of Sipelia Server
To enable the SIP protocol on Sipelia Server, you must configure the SIP port of Sipelia Server, and
ensure that all connected SIP endpoints use the same port value.
Before you begin
If deploying Sipelia, configure the system communication service.
What you should know
When configuring ports, make sure that the ports are open and that they are not being used by
another application on the same workstation. For example, if Sipelia Server is installed on the same
machine that hosts the Genetec Server, you cannot use the same port that is already being used by
Security Center or another application.
To configure the SIP port of Sipelia Server:
1 Log on to Security Center with Config Tool, and then open the Plugins task.
2 Select the Sipelia Plugin role, and then click Servers.
3 Set the following:
• SIP port: The port used to enable the SIP protocol on Sipelia Server. As a result, it is the basis
of all SIP communication in Sipelia. The default value is 5060. Every SIP endpoint, such as
softphones and SIP intercoms, that needs to connect to the Sipelia Server must have this port
value in their respective configurations.
4 If you changed the default value of the SIP port, make sure that all SIP clients that are connected to
Sipelia Server also use the new port value.
After you finish
If deploying Sipelia, define the ranges for the SIP phone extensions.
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Configuration
Defining the ranges of SIP phone extensions
Before assigning a phone extension for a given user, ring group, or SIP intercom, you can define
multiple ranges of phone extensions, and then choose a different range for each SIP entity.
Before you begin
If deploying Sipelia, configure the SIP port of Sipelia Server.
What you should know
The phone extension ranges are sets of SIP extensions from which you can assign an extension number
to each of your SIP entities. Each extension range must have a default password for the extensions, you
can only have a maximum of 1000 extensions per range, and you must have a minimum of one defined
extension range to be able to connect a SIP entity to Sipelia. By default, Sipelia provides five extension
ranges (Range 1 to Range 5), each with a default password of 1234.
To define a range of SIP phone extensions:
1 Log on to Security Center with Config Tool, and then open the Plugins task.
2 Select the Sipelia Plugin role, and then click Servers.
3 Note the extension ranges that are already defined, and decide on how to assign them to your
various SIP entities.
4 To add an extension range, click Add range ( ).
5 Enter the following:
• Start: The start value of the SIP extension range. The start value must be unique and cannot be
greater than the end value.
• End: The end value of the SIP extension range. The end value must be unique and cannot be less
than the start value.
• Description: A phrase that describes the range, and perhaps indicates what SIP entity the range
is reserved for.
• Default password: The password for every SIP extension within the range. All SIP entities whose
respective extensions lie within this range must know this password. Every SIP endpoint, such
as softphones and SIP intercoms, that needs to connect to the Sipelia Server must have this
password value (along with the extension) in their respective configurations.
• Confirm password: The confirmation of the default password. Values in both password fields
must match.
NOTE: The start and end values for the extension ranges are inclusive.
6 Click Add, and then click Apply.
Example
As shown in the following image, let's assume that you want to define an extension range reserved
solely for Security Center users, and you want to limit the number of extensions to 50. Add a unique
extension range that spans 50 numbers, and then enter a default password that applies to every
extension within this range. When configuring SIP accounts for your Security Center users, you can
assign each user an extension number within this new range, but you can only assign a maximum of 50
users to this range.
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After you finish
If deploying Sipelia, configure the recording of the audio and video of call sessions.
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Recording the audio and video of call sessions
To record the audio and video of call sessions involving users and SIP intercoms, you must configure
the recording settings provided in the Recording page of the Sipelia Plugin role.
Before you begin
Make sure that you have advanced licenses installed on your system.
What you should know
The User recording and Device recording options apply to all user and SIP intercom entities, unless an
entity is configured differently in which case it will not inherit the default values from the Sipelia Plugin
role anymore.
To record the audio and video of call sessions:
1 Log on to Security Center with Config Tool, and then open the Plugins task.
2 Select the Sipelia Plugin role, and then click Recording.
3 Enter the following:
• User recording: Enables the recording of call sessions in which user entities participate (either
as a caller or recipient of a call). Once recorded, call sessions can later be reviewed and exported
through the Call report task in Security Desk. The default value is defined here at the role level,
and is inherited by all user entities. It is possible to turn the recording on or off for only specific
user entities via the Recording audio and video setting provided in the VoIP page of the user
entity without affecting all of them.
• Device recording: Enables the recording of call sessions in which the SIP intercom entities
participate (either as a caller or recipient of a call). Once recorded, sessions can later be
reviewed and exported through the Call report task. The default value is defined here at the role
level, and is inherited by all SIP intercom entities. It is possible to turn the recording on or off for
only specific SIP intercom entities via the Recording audio and video setting provided in the VoIP
page of the SIP intercom entity without affecting all of them.
• Recording folder: The directory in which the recorded files are stored. It can be a local or a
network directory. The Sipelia Plugin role displays an error when the path format is not valid, the
local path is not accessible, or the network path does not exist or is unreachable. If a directory
specified with a local path does not exist, it will automatically be created. If a path becomes
invalid, the call session recording will stop and the recorded files will be lost.
• Automatic cleanup: Enables automatic deletion of the recorded files. This option is enabled by
default with a retention period of 30 days.
4 Click Apply.
After you finish
If deploying Sipelia, configure SIP accounts for Security Desk users.
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Configuring SIP accounts for Security Center users
To allow Security Center users to communicate with one another using the SIP-related controls in
Security Desk, you must configure a SIP account for each of your users with the appropriate privileges.
Before you begin
• Create the users that you want to configure SIP accounts for (see the Security Center Administrator
Guide for details).
• If you do not want to use the default extension ranges which are already provided, define your own
ranges of SIP phone extensions.
What you should know
Once a SIP account has been configured for a Security Center user, that user becomes a SIP entity. A
SIP entity is a Security Center entity that has SIP-related capabilities. In Security Center, examples of SIP
entities are users, ring groups, and SIP devices such as SIP intercoms.
To configure a SIP account for a Security Center user:
1
2
3
4
Log on to Security Center with Config Tool, and then open the Security task.
Click Users, and then select one from the list.
Click the VoIP tab to set up this SIP entity as a SIP endpoint.
Assign a phone extension to your SIP entity in one of the following ways:
• Click Auto-assign. Auto-assign automatically assigns the SIP entity the next available phone
extension in a given range. As a result, it is the recommended way of assigning a phone
extension to users, ring groups, and SIP intercoms. Simply click this button, choose an existing
range, and then click Apply.
• Enter the following:
• SIP extension: The SIP entity's phone extension. To be able to communicate with other
SIP endpoints, every SIP entity (user, ring group, or intercom) in Security Center must
have a unique SIP extension assigned to it. Either enter the extension manually, or use the
recommended approach of clicking Auto-assign.
• Password: The password for the extension. This password was specified when creating
the extension range. Each phone extension within a given range has its password
automatically set to match the default password for that range. Clicking Auto-assign
automatically populates this field with the correct password for the range, and is therefore
the recommended approach.
IMPORTANT: Although you can change the password for a given phone extension by simply
entering a new password, it is not recommended to do so here. It is recommended to change
passwords for phone extensions only in the Servers tab of the Sipelia Plugin role.
5 Set the following:
• Record audio and video: Allows you to record the call sessions that the SIP entity participates in
(either caller or recipient of a call). Once recorded, sessions can later be reviewed and exported
through the Call report task. The default value is inherited from the global recording settings
which are found on the Recording page of the Sipelia Plugin role. Changing this setting at the
entity level forces the entity to no longer inherit the value from the global setting, thus allowing
you to turn recording on or off for only specific entities, without affecting all of them.
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• Roaming profile: When it is turned on (default value), it stores a user's respective Security Desk
option settings in the database. As a result, users can log on to Security Desk from a different
computer on the same network and keep their settings. For example, if a user has set the option
to have incoming calls always open in a tile, this option will remain intact for this user even on a
different Security Desk workstation that is on the same network.
6 Click the Privileges tab to set up the user's privileges for Sipelia.
7 Under All privileges > Application privileges > Sipelia, select the privileges corresponding to the
actions the user is allowed to perform.
IMPORTANT: Privileges are set to Undefined by default. They must be allowed explicitly for users to
be able to make and receive calls.
8 Click Apply.
After you finish
• Associate a Security Center camera to your user.
• Allow the user to see pictures from other users.
• Add your SIP intercoms.
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Allowing users to see pictures of other users
If you want Security Center users to see the picture of other users in Sipelia, you must manually add
them in the security settings of the associated custom field.
Before you begin
• Create the Sipelia Plugin role.
What you should know
When created, the Sipelia Plugin role automatically adds the Photo custom field in Security Center. The
picture of each user, shown in the list of contacts for example, is provided by this custom field. To be
able to see the pictures, users must be added to the Security property of the custom field.
To allow a user to see the pictures:
1 Log on to Security Center with Config Tool, and then open the System task.
2 Click Custom fields, select Photo in the list, and then click Edit the item ( ).
3 Under Security, click Add an item ( ), and then select the user.
BEST PRACTICE: You can add all your Sipelia users to a single user group, and then add this group
to the custom field. This way, each time new users are added to the group, the pictures will
automatically be made visible to them.
4 Click OK > Save and close > Apply.
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Configuration
Associating Security Center cameras with users
To extend your monitoring capabilities within Security Desk, you can associate Security Center cameras
to your users, so that the live video stream from Security Center cameras is displayed during active
phone calls between users.
Before you begin
• Add your Security Center users (see the Security Center Administrator Guide for details).
• Configure SIP accounts for your Security Center users.
What you should know
Video streams from a Security Center camera do not require a SIP connection, and are therefore not
considered video calls. To configure video calls between SIP entities, you must configure the required
audio and video devices in Security Desk.
As opposed to SIP intercoms, which allow you to associate cameras, doors, zones, and output devices
to them, you can only associate a camera to a Security Center user.
To associate a Security Center camera with a user:
1
2
3
4
5
Log on to Security Center with Config Tool, and then open the Security task.
Click Users, and then select one from the list.
Click the VoIP tab to set up this SIP entity as a SIP endpoint.
In the Associated entities section, click Add an entity ( ).
In the Entity association wizard search for and select the camera entity that you want to associate
to your user, and then click Next. If you select a PTZ camera, you can select the PTZ preset whose
video stream you want to appear during an active call.
6 Confirm your selection by clicking Next, and then Apply.
7 Once the Entity association wizard has closed, click Apply to save your changes.
The entity that you selected appears in the list of associated entities.
8 (Optional) To configure the associated entity that you added, click .
9 (Optional) Turn on View events for the user. View events allows you to turn events on for users and
SIP intercoms, so that you can generate custom events for them in the system (for example, trigger
an alarm when a call is not answered). For each user or SIP intercom, you can select a custom event
for each one of the following call states: Ringing; On a call; Busy; Not answered; Error; End call. For
more information about custom events, see the Security Center Administrator Guide.
10 Associate additional cameras to your user by repeating the steps above.
The camera you have selected is linked to the user. During an active call, the live video stream from
this camera appears in the conversation window or Security Desk tile of the users with which this
user is communicating. If you have associated multiple cameras, users are able to switch between the
different video streams.
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Example
Let's assume that you have associated the following two cameras to user Charles: Front Building
Entrance and Emergency Exit. As shown in the following image, when Charles is in an active call with
Security Center users, those users can switch between the live video streams of both cameras.
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Adding SIP intercoms
To increase the security of your premises and grant access to people upon confirming their identities,
you can add a SIP intercom, and then associate it to various Security Center entities such as a camera
or door.
Before you begin
• Install the SIP intercom according to the recommendations of the intercom manufacturer.
• If you want to connect a SIP intercom to a SIP trunk and not Sipelia Server, add a SIP trunk, and then
define associated dial plan rules.
• Define a range of phone extensions for your SIP intercoms. It is recommended to dedicate extension
ranges for each particular SIP entity, especially intercoms that communicate with Sipelia Server
through a SIP trunk.
What you should know
A SIP intercom is an intelligent SIP endpoint that provides two-way phone connectivity in a SIP
environment. In Security Center, a SIP intercom is one of the established SIP entities, and it is the only
SIP entity that is an actual device. The other SIP entities are Security Center users and ring groups.
To add a SIP intercom:
Log on to Security Center with Config Tool, and then open the Plugins task.
Select the Sipelia Plugin role.
At the bottom of the page, click Add intercom ( ).
Enter a descriptive name for your SIP intercom, and then click Add.
The Logical view task opens, and the intercom you added appears in the list of entities.
5 Click the VoIP tab to set up this SIP entity as a SIP endpoint.
6 Assign a phone extension to your SIP entity in one of the following ways:
1
2
3
4
• Click Auto-assign. Auto-assign automatically assigns the SIP entity the next available phone
extension in a given range. As a result, it is the recommended way of assigning a phone
extension to users, ring groups, and SIP intercoms. Simply click this button, choose an existing
range, and then click Apply.
• Enter the following:
• SIP extension: The SIP entity's phone extension. To be able to communicate with other
SIP endpoints, every SIP entity (user, ring group, or intercom) in Security Center must
have a unique SIP extension assigned to it. Either enter the extension manually, or use the
recommended approach of clicking Auto-assign.
• Password: The password for the extension. This password was specified when creating
the extension range. Each phone extension within a given range has its password
automatically set to match the default password for that range. Clicking Auto-assign
automatically populates this field with the correct password for the range, and is therefore
the recommended approach.
IMPORTANT: Although you can change the password for a given phone extension by simply
entering a new password, it is not recommended to do so here. It is recommended to change
passwords for phone extensions only in the Servers tab of the Sipelia Plugin role.
7 Set the following:
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• Record audio and video: Allows you to record the call sessions that the SIP entity participates in
(either caller or recipient of a call). Once recorded, sessions can later be reviewed and exported
through the Call report task. The default value is inherited from the global recording settings
which are found on the Recording page of the Sipelia Plugin role. Changing this setting at the
entity level forces the entity to no longer inherit the value from the global setting, thus allowing
you to turn recording on or off for only specific entities, without affecting all of them.
8 Click Apply.
9 Add additional SIP intercoms by repeating the steps above.
After you finish
• Associate a Security Center entity with your SIP intercom.
• Register your SIP intercom.
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Configuration
Associating Security Center entities with SIP intercoms
To confirm caller identification through live video and grant access to the premises through access
controlled doors, you can associate various Security Center entities to a SIP intercom, and then control
everything through Security Desk.
Before you begin
Add and configure your SIP intercom in Config Tool.
What you should know
With SIP intercoms, you can associate cameras, doors, zones, and device outputs along with their
respective output behaviors.
To associate a Security Center entity to a SIP intercom:
1
2
3
4
Log on to Security Center with Config Tool, and then open the Logical view task.
From the logical view, select a SIP intercom, and then click the VoIP tab.
In the Associated entities section, click Add an entity ( ).
In the Entity association wizard, select one of the following entities, and then click Next. Follow the
onscreen instructions to complete the entity association.
• Camera: The camera that you want to associate with the SIP entity. You can associate a camera
so that the live video stream from this Security Center camera is displayed during active phone
calls between SIP entities.
• Door: The door that you want to associate with the SIP intercom. You can associate a door so
that you can unlock it to grant access to callers that use the intercom, and then lock it once the
callers have entered.
• Zone: The zone that you want to associate with the SIP intercom. You can associate a zone so
that you can arm and disarm it based on the callers that use the intercom. As a result, you can
grant or deny callers access to a particular section of your premises. Once set up, in the Security
Desk conversation window or tile, click to disarm the zone.
5
6
7
8
• Device: The output of a device that you want to associate with the SIP intercom. Each output
must have an associated output behavior; for example, sounding a buzzer (via an output relay)
when a window that is equipped with a glass break sensor (connected to an input) is shattered.
Once set up, you can trigger the selected output behavior during a call.
If you selected a device output, do the following to define the output behavior:
a) In the Alias field, enter a short, descriptive name that allows you to quickly identify the output
behavior.
b) Select an applicable output type (Normal, Active, or a custom one). You can create multiple
custom output types based on the State, Pulse, and Periodic types. For more information on
output behaviors, see the Security Desk Administrator Guide.
Once the Entity association wizard has closed, click Apply to save your changes.
The entity that you selected appears in the list of associated entities.
(Optional) If your SIP intercom has a built-in camera, and you want the video stream from this
camera to be displayed in conversations involving the intercom, you can associate that built-in
camera by clicking .
(Optional) To configure the associated entity that you added, click .
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9 (Optional) Turn on View events for the SIP intercom. View events allows you to turn events on
for users and SIP intercoms, so that you can generate custom events for them in the system
(for example, trigger an alarm when a call is not answered). For each user or SIP intercom, you
can select a custom event for each one of the following call states: Ringing; On a call; Busy; Not
answered; Error; End call. For more information about custom events, see the Security Center
Administrator Guide.
10 Repeat the above steps to associate additional entities to the SIP intercom.
11 For multiple entities of the same type, use the arrow buttons in the Associations panel to define
the order in which the entities appear within the conversation window or tile in Security Desk.
For example, if you have associated multiple cameras, the live video stream from the camera that
appears first in the list is the default stream. Users can switch to the video streams of other cameras
by clicking , and then selecting another camera.
If you associate all of the possible entities with a SIP intercom, all of the entities appear within the
conversation window or tile in Security Desk. As shown in the following image, each entity can be
controlled during an active call.
• A: Trigger the output behavior of a device
• B: Arm and disarm a zone
• C: Open and close a door
• D: View the video stream of the selected camera
Example
Let's assume that you want to facilitate how operators respond to lost card requests. With a SIP
intercom installed at the main entrance of the building, you can associate the following two entities
with that intercom: the camera at the main entrance, and the door at the main entrance. When Charles
calls from that SIP intercom, claiming that he has lost his card and cannot enter the building, you can
confirm Charles' identity by comparing the live video against his cardholder profile, and then granting
or denying access to the building accordingly.
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Registering your SIP intercom with Sipelia Server
To make calls from your SIP intercom to Security Center users, you must register the intercom with
Sipelia Server.
Before you begin
• Install the SIP intercom according to the recommendations of the intercom manufacturer.
• Add and configure your SIP intercom in Config Tool.
What you should know
Because there are a variety of SIP intercoms that you can install, the way they are configured and
registered to Sipelia Server might differ. The steps below provide a general overview of the settings
that must be configured. Always refer to the documentation provided by the manufacturer of the SIP
intercom for information on how to configure and register the intercom.
To register your SIP intercom with Sipelia Server:
• Create a SIP account on your intercom.
• Enter an applicable name for your SIP account. This name can be the same as the one given to the
intercom when adding it to Security Center, but it does not need to be. The SIP account name is not
used during SIP communications.
• Enter the domain or IP address of Sipelia Server.
You can find the IP address of Sipelia Server in Config Tool > Network view > Properties.
• For the SIP port value, enter the value you have configured. The default value used by the Session
Initiation Protocol (SIP) is 5060.
• Enter the phone extension which was assigned to this intercom in Security Center. In certain SIP
clients, you must enter the extension number as the username.
• Enter the password for the phone extension that has been assigned to the intercom.
• Register the intercom so that it can communicate with Sipelia Server.
The SIP intercom is ready to make and receive calls.
After you finish
If deploying Sipelia, create basic ring groups.
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Ring groups
A ring group is a group of SIP entities that has its own unique phone extension. All entities (or
members) within a ring group are part of a call list, and all members get called when the ring group
extension is called. The members of a ring group can either be called all at once, or successively at a set
interval. The call stops ringing once any one of the members within a call list answers the call.
In Security Center, there are two types of ring groups: basic and custom.
Basic ring groups
A basic ring group is a Security Center user group which has been assigned its own unique extension. In
a basic ring group, you can only include Security Center users and other Security Center user groups.
A basic ring group has the following characteristics:
• It is a Security Center user group.
• It can only include Security Center users and other Security Center user groups.
• It can be assigned its own unique SIP extension.
• Once the extension is dialed, all members of the user group set with a SIP extension are called.
• To be able to receive a call, users that are part of a basic ring group must have their own dedicated
SIP extension.
• If users do not have a SIP extension, they are bypassed when the ring group that they are a part of is
called.
Custom ring groups
A custom ring group is a ring group that can include any combination of the following entities: users,
user groups, and SIP devices. Whereas basic ring groups can only include users and user groups, custom
ring groups can also include SIP devices.
A custom ring group has the following characteristics:
• It can include any combination of the following entities: users, user groups, and SIP devices.
• It can be assigned its own unique SIP extension.
• To be able to receive a call, users and SIP devices within a custom ring group must have their own
unique SIP extensions.
• If users and SIP devices do not have a SIP extension, they are bypassed when the custom ring group
that they are a part of is called.
• Security Center user groups that are part of a custom ring group do not need to have their own
unique extensions, but each user entity included in the custom ring group must provide one. User
entities that do not have a SIP extension will be omitted when the group is called.
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Creating basic ring groups
To call multiple Security Center users at the same time, you can create a basic ring group by simply
assigning a phone extension to a Security Center user group.
Before you begin
Configure SIP accounts for your Security Center users.
What you should know
A basic ring group is a Security Center user group which has been assigned its own unique extension. In
a basic ring group, you can only include Security Center users and other Security Center user groups.
To create a basic ring group:
1 Log on to Security Center with Config Tool, and then open the Security task.
2 Click User groups, and then select one from the list.
3 Create a Security Center user group that contains all of the users you want as part of your ring
group. For information on creating user groups in Config Tool, see the Security Center Administrator
Guide.
4 Click the VoIP tab.
The members of the user group appear in the Call list field.
5 Assign a phone extension to your ring group in one of the following ways:
• Click Auto-assign. Auto-assign automatically assigns the SIP entity the next available phone
extension in a given range. As a result, it is the recommended way of assigning a phone
extension to users, ring groups, and SIP intercoms. Simply click this button, choose an existing
range, and then click Apply.
• Enter the following:
• SIP extension: The SIP entity's phone extension. To be able to communicate with other
SIP endpoints, every SIP entity (user, ring group, or intercom) in Security Center must
have a unique SIP extension assigned to it. Either enter the extension manually, or use the
recommended approach of clicking Auto-assign.
6 In the Call recipients field, select one of the following:
• All at once: All members of a call list are called at the same time. The call stops ringing once any
one of the members within a call list answers the call.
• Successively every: Members get called one after another in sequence, with a set delay between
each call. The order in which the members are called is based on the order in which they appear
in the call list. This sequence of calls is repeated until any one of the members within a call list
answers the call. If a member declines the call, the next member in the call list is immediately
called, regardless of whether the set delay between calls has elapsed. The minimum delay is
10 seconds. Because it might affect how long a call to a ring group goes unanswered, it is not
recommended to set a delay that is too high.
7 (Optional) To change the order in which the members of the ring group are called, use the arrow
buttons to move the members up or down. This is available only when Successively every is
selected.
8 Click Apply.
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Example
As shown in the image below, this basic ring group includes five Security Center users, each with their
own dedicated extensions. The call sequence for this ring group is set at Successively every 10 seconds.
As a result, when the ring group extension (7002) is called, Charles' extension (6001) is called first. If
Charles does not answer or decline the call within 10 seconds, Clive's extension (6002) is called. The
same sequence continues until one of the members in the call list answers the call.
After you finish
To create ring groups that include other ring groups and SIP intercoms, create custom ring groups.
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Creating custom ring groups
To call multiple SIP entities at the same time, you can create a custom ring group in Config Tool.
Before you begin
Depending on which entities you want to include in your custom ring group, do one or more of the
following:
• Configure SIP accounts for your Security Center users.
• Create a basic ring group.
• Add your SIP intercoms.
What you should know
A custom ring group is a ring group that can include any combination of the following entities: users,
user groups, and SIP devices. Whereas basic ring groups can only include users and user groups, custom
ring groups can also include SIP devices.
To create a custom ring group:
1
2
3
4
5
Log on to Security Center with Config Tool, and then open the Plugins task.
Select the Sipelia Plugin role, and then click Ring groups.
Click Add ring group ( ).
Enter a descriptive name for your custom ring group, and then click Add.
Assign a phone extension to your ring group in one of the following ways:
• Click Auto-assign. Auto-assign automatically assigns the SIP entity the next available phone
extension in a given range. As a result, it is the recommended way of assigning a phone
extension to users, ring groups, and SIP intercoms. Simply click this button, choose an existing
range, and then click Apply.
• Enter the following:
• SIP extension: The SIP entity's phone extension. To be able to communicate with other
SIP endpoints, every SIP entity (user, ring group, or intercom) in Security Center must
have a unique SIP extension assigned to it. Either enter the extension manually, or use the
recommended approach of clicking Auto-assign.
6 In the Call list field, click Add entity ( ).
7 In the Select entity dialog box, choose the different entities that you want to include in your custom
ring group. As shown in the following image, you can use the Entity type drop-down list to filter by
entity type, and then search for the entities by name.
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8 Once you have selected all of your entities, click Select.
9 (Optional) To configure the associated entity that you added, click
10 In the Call recipients field, select one of the following:
.
• All at once: All members of a call list are called at the same time. The call stops ringing once any
one of the members within a call list answers the call.
• Successively every: Members get called one after another in sequence, with a set delay between
each call. The order in which the members are called is based on the order in which they appear
in the call list. This sequence of calls is repeated until any one of the members within a call list
answers the call. If a member declines the call, the next member in the call list is immediately
called, regardless of whether the set delay between calls has elapsed. The minimum delay is
10 seconds. Because it might affect how long a call to a ring group goes unanswered, it is not
recommended to set a delay that is too high.
11 (Optional) To change the order in which the members of the ring group are called, use the arrow
buttons to move the members up or down. This is available only when Successively every is
selected.
12 Click Apply.
Example
As shown in the image below, this custom ring group includes three entities: one user, one user group,
and one basic ring group. The call sequence for this custom ring group is set at Successively every 10
seconds. As a result, when the ring group extension (5001) is called, Charles' extension (6001) is called
first. If Charles does not answer or decline the call within 10 seconds, the users that make up the Plan
Manager operators user group, and that already have an assigned extension, are each called. The same
sequence continues until one of the members in the call list answers the call.
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After you finish
If deploying Sipelia, configure your audio and video devices, so that Security Center users can make and
receive voice and video calls.
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Configuring devices for voice and video calls
To participate in voice and video calls, you can connect the required audio and video equipment to
the Security Desk workstations onto which Sipelia Client is installed, and then configure relevant
configuration settings for each Security Center user.
Before you begin
• Configure SIP accounts for your Security Center users.
• Install Sipelia Client on each of the Security Desk workstations that run Sipelia.
• Install the required headsets and webcams. For optimal audio quality, it is recommended to use
headsets instead of microphones and speakers.
To configure devices for voice and video calls:
1 Log on to Security Center with Security Desk.
2 Click Options > Sipelia.
3 In the Audio and video section, select the physical audio and video devices that are used for calls.
IMPORTANT: Make sure that the devices are properly connected to the Security Desk workstations
that run Sipelia.
4 Click to expand the Advanced section, and set the following settings, as required:
• Video codecs: The video codecs that are supported by Security Desk for video communication.
By default, the H.264 and H.263 codecs are turned on, and should suffice for most cases. As
a result, it is recommended to keep the default settings, and to be aware that changing video
codecs can disrupt the video that is streamed during video calls. To be able to view video during
a SIP video call, the SIP clients that are involved in a call must all support at least one common
video codec. For example, if SIP client A only supports the H.264 codec and SIP client B only
supports H.263, no video is streamed during a call session between the two SIP clients.
• UDP port range: The port range for the User Diagram Protocol (UDP). The UDP ports are used
by the different SIP clients to send and receive communication data. The default range is from
20000 to 20500. It is recommended to keep the default settings, and to change them only if
Sipelia logs any port-related issues about making or receiving calls with Security Desk.
5 Set the following call-related options, as required:
• Open new calls in: Select whether you want all incoming calls to automatically open in the
conversation window or in a tile within the Monitoring task in Security Desk.
• Play sound on new call: Select this check box if you want to hear an audible ring when receiving
an incoming call.
6 Repeat these steps on each of the Security Desk workstations that run Sipelia.
After you finish
• Test the audio and video devices by making voice and video calls between SIP clients.
• If deploying Sipelia, configure two-way communication between Sipelia Server and other SIP
servers.
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Configuring two-way communication between Sipelia
Server and other SIP servers
To expand your SIP capabilities, you can configure two-way communication between Sipelia Server and
other SIP servers, so SIP extensions from either server can call each other.
Before you begin
• Read the documentation provided by the manufacturer of the SIP server that you want to connect
to Sipelia Server.
NOTE: For the purpose of this topic, we'll refer to the SIP server that you want to connect to as SIP
Server A.
• Make sure that SIP Server A supports trunking and is able to connect to other SIP servers (in
our case, Sipelia Server). Some SIP servers do not support trunking, and therefore, cannot call
extensions registered to other SIP servers.
• Make sure that SIP Server A has been certified by Genetec as a compatible hardware component,
and that it is included in Sipelia's supported hardware list for SIP servers and intercom servers.
What you should know
To have a SIP environment like the one shown in the following image, you must properly configure
two-way communication between the two SIP servers (Sipelia Server and SIP Server A which, for
example, is embedded within an intercom server). In this illustration, Sipelia Server must add SIP Server
A as a SIP trunk, and SIP Server A also needs to add and configure Sipelia Server as a SIP trunk. Also,
each SIP server must define appropriate dial plans to be able to contact the other SIP server.
SIP Server A
Sipelia Server
SIP intercom registered to SIP Server A
Operator using Sipelia-based functions
in Security Desk
Because there are many different SIP servers available on the market, and numerous possible
implementations for each, the method used to configure a SIP trunk and dial plan process differs for
each SIP server. As a result, the steps listed below only provide general instructions on how to set up
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Configuration
a two-way communication between Sipelia Server and SIP Server A. However, in this topic, and in the
related topics about SIP trunks and dial plans, specific instructions are provided for how to set up a
one-way communication between Sipelia Server and the SIP trunk for SIP Server A. This way, extensions
on Sipelia Server can call those on SIP server A.
To connect Sipelia Server to SIP Server A:
1 Add SIP Server A as a SIP trunk.
2 Define dial plan rules that allow SIP extensions registered to Sipelia Server to call SIP extensions
registered to SIP Server A.
3 Import the dial plan rules in Config Tool.
To connect SIP Server A to Sipelia Server:
1 Add Sipelia Server as a SIP trunk. Refer to the documentation of SIP Server A for details on how to
add SIP trunks.
2 Define and implement dial plan rules that allow SIP extensions registered to SIP Server A to call SIP
extensions registered to Sipelia Server. Refer to the documentation of SIP Server A for details on
how to define and implement dial plan rules.
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Configuring your SIP intercom to call a specific extension
When your SIP intercom can only be configured to call a specific extension when a button is pressed,
it is recommended to configure your SIP intercom to call the SIP extension of a ring group, so that you
can manage the list of users who will receive the call.
Before you begin
• Install the SIP intercom according to the recommendations of the intercom manufacturer.
• Add and configure your SIP intercom in Config Tool.
What you should know
When your SIP intercom only allows the configuration of one SIP extension, it is recommended that you
use the extension of a ring group. With a ring group you can then decide who will be called when the
button is pressed on the SIP intercom and it calls the extension. The ring group also provides you with
the flexibility to adjust the list of recipients to meet your needs.
Because there are a variety of SIP intercoms that you can install, the way they are configured and
registered to Sipelia Server might differ. The steps below provide a general overview of the settings
that must be configured. Always refer to the documentation provided by the manufacturer of the SIP
intercom for information on how to configure and register the intercom.
To configure your SIP intercom to call the SIP extension of a ring group:
1
2
3
4
Add and configure a ring group in Config Tool.
Take note of the SIP extension that you assigned to the ring group.
Create a SIP account on your intercom.
Enter an applicable name for your SIP account. This name can be the same as the one given to the
intercom when adding it to Security Center, but it does not need to be. The SIP account name is not
used during SIP communications.
5 Enter the domain or IP address of Sipelia Server.
You can find the IP address of Sipelia Server in Config Tool > Network view > Properties.
6 For the SIP port value, enter the value you have configured. The default value used by the Session
Initiation Protocol (SIP) is 5060.
7 Enter the SIP extension which was assigned to the ring group in Security Center. In certain SIP
clients, you must enter the extension number as the username.
The SIP intercom will call the extension of the ring group when its button is pressed. This means that
one or more recipients could be called all at once or one after another according to the ring group
configuration.
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Adding SIP intercom icons to a Plan Manager map
In a system where Sipelia and Plan Manager are installed, you can add SIP intercom icons to your maps
so that they are displayed in Security Desk and can be used by the operators to make and receive calls.
Before you begin
• Have Plan Manager deployed on your system. For more details on how to deploy, configure and
operate Plan Manager, refer to the Plan Manager User Guide.
To add a SIP intercom icon to a Plan Manager map:
1 Log on to Security Center with Security Desk and open a Monitoring task.
2 From the Logical view, double-click a Plan Manager map ( ) or drag the map to a tile of your
choice.
3 Navigate to the location on the map where you want to add the SIP intercom icon.
4 In the Plan Manager tile, select the Edit ribbon.
A thick red border appears around the Plan Manager workspace.
5 From the Logical view, click the SIP intercom you want and drag it to the required location on the
map.
6 Adjust the size, position, and orientation of the SIP intercom icon with the mouse.
7 In the panel displayed on the left, configure the SIP intercom icon properties.
The properties are grouped by categories. Click a group heading to open it.
• Identity: Name of the SIP intercom icon as it will be displayed on the map.
• States: Sets of properties (image, image size, color, transparency, blink rate) representing each
possible state of the SIP intercom icon on the map.
• Position: Properties affecting the general appearance of the SIP intercom icon on the map.
• Associated entity: List of entities linked to the SIP intercom.
8 From the Edit ribbon, click Save changes( ).
After you finish
Click the Home ribbon and try to make and receive call using the SIP intercom icon you just added.
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Configuring a network interface with the highest priority
To ensure that the communication between Security Desk and Sipelia Server works properly when the
computer on which Security Desk is running provides multiple network interfaces (cards), you must
configure the network interface to be used for Sipelia with the highest priority.
What you should know
Applications such as VPN can automatically modify the network interface priorities in Windows, when
establishing a connection for example. In this case you may need to reconfigure the priorities again
after you used those applications, or try to have a computer dedicated to Security Desk.
To configure a network interface with the highest priority:
1
2
3
4
5
In the Windows Control Panel, click Network and Internet > Network and Sharing Center.
On the left pane, click Change adapter settings.
Press ALT to display the top menu, and then click Advanced > Advanced settings.
In the Advanced settings configuration window, click Adapters and Bindings.
Under Connections, select the network interface that must be used by Sipelia and move it to the
top of the list using the arrows provided on the right.
The network interface displayed at the top of the list is configured with the highest priority.
6 Click OK to confirm the changes.
7 To check that the changes were saved properly, open a command prompt, type ipconfig, and
then press ENTER.
8 Verify that the order of the network interfaces listed in the command prompt is the same as the one
configured in Adapters and Bindings.
The network interface selected for Sipelia is configured with the highest priority in Windows and will be
used by Security Desk to connect and to register to Sipelia Server.
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4
SIP trunks and dial plans
This section includes the following topics:
•
"Adding SIP trunks" on page 46
•
"Dial plans" on page 47
•
"Regular expressions in Sipelia" on page 49
•
"Defining dial plan rules" on page 51
•
"Importing dial plans" on page 52
•
"Dial plan scenario 1: Forwarding to a SIP trunk all calls starting with a prefix" on page 53
•
"Dial plan scenario 2: Reserving a range of SIP extensions for local calls" on page 55
•
"Dial plan scenario 3: Reserving a range of SIP extensions for calls to a SIP trunk" on page 58
•
"Dial plan scenario 4: Replacing source SIP extensions" on page 61
•
"Dial plan scenario 5: Removing prefix on source SIP extensions from a SIP trunk" on page 63
•
"Dial plan scenario 6: Forwarding calls to another SIP extension on schedule" on page 65
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Adding SIP trunks
To connect to SIP servers other than Sipelia Server, you can add a SIP trunk in Config Tool, and then
define applicable dial plan rules to make and receive calls between the SIP servers.
What you should know
SIP trunks work together with dial plans. For example, to connect Sipelia Server to another SIP server
(SIP Server A) through a SIP trunk, you must define dial plan rules that instruct Sipelia Server about
which calls to reroute through the trunk and on towards the new SIP extension destinations that are
registered to SIP Server A. Furthermore, if you want proper two-way communication between Sipelia
Server and SIP Server A, both SIP servers need to be configured accordingly.
To add a SIP trunk:
Log on to Security Center with Config Tool, and then open the Plugins task.
Select the Sipelia Plugin role, and then click SIP trunks.
Click Add SIP trunk ( ).
In the Name field, enter a descriptive name for your SIP trunk. The name of the SIP trunk must be
unique, because it will be used in the dial plan rules involving this SIP trunk.
Example: For the purposes of this example, let's name the SIP trunk TrunkSIPServerA for SIP Server
A.
5 Enter the following:
1
2
3
4
• IP address: The IP address of the SIP server that you want to connect Sipelia Server to.
• SIP port: The port used by the SIP trunk to communicate with the Sipelia Server. Because SIP
trunks are SIP servers, the default value is 5060.
Example: Let's assume that the IP address of TrunkSIPServerA is the following: 10.150.4.100. And, as
stated above, the default port is 5060.
6 Click Add, and then click Apply.
The SIP trunk TrunkSIPServerA has been added.
After you finish
To call extensions registered on TrunkSIPServerA, define the applicable dial plan rules.
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Dial plans
A dial plan is a collection of rules that defines how calls are routed locally or between two SIP trunks.
Dial plans ensure that calls are routed and rerouted correctly, and they also allow administrators to
block calls to certain geographic locations or ensure the privacy of the callers.
A dial plan defines one or multiple rules to specify the following:
• How calls can reach SIP extensions that reside on other SIP servers.
• How calls coming from other SIP servers can reach SIP extensions that reside on Sipelia Server.
• How calls can be forwarded even when they remain local to Sipelia Server.
• How Sipelia Server can modify dialing information such as source or destination SIP extensions
during call routing.
For example, if a SIP extension registered to Sipelia Server needs to call an extension that resides on SIP
Server A, you must first configure a SIP trunk in Sipelia Server to connect to SIP Server A. Then you must
define a dial plan to route the call from Sipelia Server to SIP Server A. In Sipelia, dial plans can also be
used to route local calls, that is, calls that must remain in Sipelia Server.
NOTE: Dial plans also need to be configured on SIP Server A to route calls from and to Sipelia Server.
Refer to the documentation provided by the manufacturer of SIP Server A for information on how to
configure dial plans.
Dial plan rules
Dial plan rules are stored in comma-separated values (CSV) text files, and these files can contain
multiple rules. Each rule appears on a separate line and each comma-separated value (or value field)
of a rule is separated by a semicolon (;). Although CSV files support different types of separators (or
delimiters), dial plan rules in Sipelia only support the semicolon (;) separator. You can define dial plan
rules for a number of reasons. Most commonly, dial plan rules are used to call SIP extensions that are
registered to SIP servers other than Sipelia Server, and to redirect local calls from extensions residing
within Sipelia Server.
Each dial plan rule must define the values as shown in the following example.
RouteCallToTrunkSIPServerA;00000000-0000-0000-0000-000000000006;ON;
local;TrunkSIPServerA;(.*);^780(.*);(.*);\1;
• Name (A): The name of the dial plan rule. If the name is not provided, a default value is used upon
import (DialplanRule1). Enter a name that will help you remember the purpose of the rule.
• Schedule (B): The schedule that defines when this rule must be in effect. The long string of digits
for this value represent a GUID (Globally Unique Identifier). If this GUID is not provided, or it is
incorrect, the default value of Always is used upon import.
TIP: The schedule can be changed graphically from the Dial plans page in Config Tool once the dial
plan rule has been imported, so you can leave it empty here and let the system assign the default
schedule.
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• Status (C): The status of the rule (On = Active; Off = Inactive). This value is not case sensitive. If this
value is not provided, the default value Off is used upon import.
• Direction from (D): The SIP server from where the call is originating (the source server). Possible
values are local or the name of a SIP trunk. Sipelia Server is always the local SIP server. If a SIP
trunk name is entered here and does not exist in the Sipelia Server configuration, a warning will be
displayed and this value will revert to local during import.
• Direction to (E): The SIP server that receives the call (the destination server). Possible values are
local or the name of a SIP trunk. The name of the SIP trunk must be unique, and must match the
name that was entered in the SIP trunks tab when adding the SIP trunk. Sipelia Server is always
the local SIP server. If a SIP trunk name is entered here and does not exist in the Sipelia Server
configuration, a warning will be displayed and this value will revert to local during import.
TIP: After adding the SIP trunk in Config Tool, you can copy its name by right-clicking on the
name. It is recommended to copy the name, and then paste it within your dial plan rule to avoid
misspelling it.
• Source (F): The regular expression that identifies the extension of the caller (the source caller).
The dial plan rule is only applied if there is a match between the regular expression and the caller's
extension. This field is mandatory.
• Destination (G): The regular expression that identifies the extension of the recipient (the
destination recipient). The dial plan rule is only applied if there is a match between the regular
expression and the recipient's extension. This field is mandatory.
• New source (H): The value that changes the caller's extension (if the rule is applied). For example, if
the caller at extension 1001 calls the recipient at extension 2001, and the New source value is 4001,
the recipient at extension 2001 sees that the incoming call is coming from extension 4001, not 1001.
Possible values can also include regular expressions.
• New destination (I): The SIP extension that is receiving the call (if the rule is applied). Possible
values can also include regular expressions.
Dial plan rules priority
Rules defined in a dial plan are listed in order of priority. A rule located in a previous row is considered
to be a higher priority rule. If a call matches two or more rules, the first rule listed in the dial plan will
always apply. The order of priority can be changed after they were imported in Config Tool using the
arrows located on the right side of the list.
Sample scenarios
For more information on how to use regular expressions in dial plan rules, you can see the following
sample scenarios:
• Forwarding to a SIP trunk all calls starting with a prefix.
• Reserving a range of SIP extensions for local calls.
• Reserving a range of SIP extensions for calls to a SIP trunk.
• Replacing source SIP extensions.
• Removing the prefix on source SIP extensions from a SIP trunk.
• Forwarding calls to another SIP extension on schedule.
In the Sipelia installation folder, typically C:\Program Files (x86)\Genetec Sipelia, the Samples folder
contains a zip file from which you can get the sample dial plan files associated with the above
scenarios.
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Regular expressions in Sipelia
A regular expression is a sequence of symbols used by a regular expression engine to identify all the
strings of characters that match a specific search pattern without having to list all the possible discrete
values that must be returned. The Microsoft's .NET Framework Regular Expression engine is the engine
used in Sipelia.
In Sipelia, regular expressions are used in dial plan rules to do the following:
• Look for specific SIP extensions from which calls are made.
• Look for destination SIP extensions to which calls are made.
• Look for specific prefixes added to SIP extensions.
• Change SIP extensions from which calls are received.
• Change SIP extensions to which calls are sent.
Regular expression elements
Regular expressions used in dial plan rules typically include the following elements:
• (.*): Match any value.
• Use this element in the Source (F) or Destination (G) fields to request Sipelia Server to look for
any source or destination extension when calls are made.
• Use this element in the New source (H) or New destination (I) fields to ensure that the source or
destination extension will remain unchanged when routed by Sipelia Server.
• \n: Match the value of a capturing group from a previous related field. A capturing group is one or
more regular expression elements that are typically specified between parentheses and match a
specific pattern.
• Use this element in the New source (H) field to specify the capturing group from the Source field
to be used as the new source value. n represents the ranking of the capturing group (\1; \2; \5,
etc.) of the regular expression. For example, if Source contains 550[1-5](.*), enter \2 to use (.*)
as the New source value, which in this case will remove the prefix 550 and the following digit.
• Use this element in the New destination (I) field to specify the capturing group from the
Destination field to be used as the new destination value.
• n: Match a specific value. In Sipelia, this value generally represents a specific SIP extension.
• Use this element in the New source (H) field to specify a different extension number from which
the calls are made. This is useful if you want calls to come from an extension which differs from
the original source extension.
• Use this element in the New destination (I) field to specify the extension number that will
receive the calls. This is useful if you want to forwards calls to an extension which differs from
the original destination extension.
• [first - last]: Match any single character in the range from first to last.
• {n}: Match the previous element exactly n times.
• \b: Use at the beginning and at the end of a series of regular expression elements to match on
whole word only (not only a part of it).
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Example
For example, the regular expression \b6[0-9]{2}\b could be used to look for SIP extensions 600 to 699.
Regular expression construct Description
\b
SIP extension boundary. Used together with the same \b element at
the end of the expression, this requires the whole SIP extension to be
matched. Starting or ending characters will not be omitted.
6
Look for a SIP extension that begins with 6.
[0-9]
Look for the digits 0 through 9.
{2}
Look for 2 occurrences of the above digits following the 6.
\b
Match a SIP extension boundary.
For more information on regular expressions, see Microsoft's website.
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Defining dial plan rules
To call SIP extensions registered to SIP servers other than Sipelia Server, or simply redirect calls from
within Sipelia Server, you must define dial plan rules, and then import the rules into Config Tool.
Before you begin
• If you want to call SIP extensions that are registered to other SIP servers, add a SIP trunk for that SIP
server.
• Familiarize yourself with dial plans and dial plan rules used in Sipelia.
What you should know
Rules defined in a dial plan are listed in order of priority. A rule located in a previous row is considered
to be a higher priority rule. If a call matches two or more rules, the first rule listed in the dial plan will
always apply. The order of priority can be changed after they were imported in Config Tool using the
arrows located on the right side of the list.
To define a dial plan rule:
1 Create a new text file for your dial plan with the .txt or .csv extension, or open an existing one.
2 Write an applicable dial plan rule. Depending on what your dial plan rule is for, you can base your
rule on a sample scenario or create a rule of your own.
3 Enter a value for each field required by the rule.
IMPORTANT: A dial plan rule must contain the correct number of comma-separated values for it to
be imported successfully. Each comma-separated value of a rule must be separated by a semicolon
(;).
4 Repeat the same steps for additional rules that you require in your plan.
5 Save and close the dial plan file.
After you finish
Import your dial plan rule in Config Tool.
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Importing dial plans
Once dial plan rules have been defined, you must import the dial plan files into Config Tool so that the
rules can be applied accordingly.
Before you begin
• Add a SIP trunk.
• Define applicable dial plan rules.
What you should know
If you delete ( ) a dial plan rule that you have imported, the rule is no longer available to the dial plan
system of Sipelia Server. If you want the deleted rule to be available again, you must reimport the dial
plan file which contains the rule. If you want the dial plan system to no longer apply a rule, simply make
the rule inactive, and keep it as part of your imported rules in case you ever want to reapply the rule
again.
When updating a dial plan rule that has already been imported once, you do not need to delete the
existing rule from the Dial plans page. In the dial plan file, simply make the necessary changes to the
rule, make sure not to change the name of the rule, and then reimport the rule. Upon import, dial plan
rules with new names get added to the list of rules; those whose names are already part of the list, are
simply updated.
To import a dial plan:
1
2
3
4
Log on to Security Center with Config Tool, and then open the Plugins task.
Select the Sipelia Plugin role, and then click Dial plans.
Click Import dial plan rules from file ( ).
Select the dial plan file, and then click OK.
The dial plan is imported and the rules contained within your file appear on separate lines on the
Dial plans page.
IMPORTANT: If a dial plan rule does not have the correct number of comma-separated values, the
rule is not imported and an error is generated.
5 If your dial plan file contains issues, click the Show errors and warnings button to open the Parser
result window.
This window classifies the issues by type (error or warning), and provides information on each issue.
Warnings do not need to be corrected, but errors do. Simply modify the dial plan file accordingly,
and then reimport the file.
6 For each dial plan rule that you import, you can edit the following:
• Schedule: The Security Center schedule that defines when this rule must be in effect. The rule is
applied only if the schedule condition is met.
• Status: The status of the rule. Only Active rules are applied.
7 (Optional) To change the order in which the dial plan rules are applied, use the arrow buttons to
move the rules up or down.
8 Click Apply.
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Dial plan scenario 1: Forwarding to a SIP trunk all calls
starting with a prefix
In this scenario, a dial plan rule is used to route to a SIP trunk any call that begins with a specific prefix.
Scenario
• What do you want to do? Your SIP extension is registered to Sipelia Server and you need to connect
to SIP Server A because you want to call SIP extensions (SIP client, SIP intercom, etc.) that are
registered to that SIP trunk. Doing so allows you to establish a one-way communication with SIP
Server A.
• What you need to do before defining the rule? Add a SIP trunk in Config Tool for SIP Server A and
name it TrunkSIPServerA. This trunk name must be unique, so that the dial plan rule that you will
create will not conflict with other rules, and your rule will be applied correctly.
• How can you do it? To route calls to TrunkSIPServerA, you can use a dialing prefix; for example, the
prefix 780. This means that SIP extensions on Sipelia Server must dial this prefix if they want to call
SIP extensions on TrunkSIPServerA. The prefix approach mimics that of dialing 9 to get an outside
line in a typical PBX system. The prefix can be any number that you want, as long as it is defined in
your dial plan rule.
Example of the dial plan rule
The following is an example of a dial plan rule that accomplishes the above mentioned scenario.
Scenario1-PrefixToTrunk ;00000000-0000-0000-0000-000000000006;ON;
local;TrunkSIPServerA;(.*);^780(.*);(.*);\1;
The value labels identified below correspond to the column labels which appear on the Dial plans
page of the Sipelia Plugin role.
Value
letter
Value label
Description
A
Name
The name of the rule indicates that calls made from Sipelia Server
and starting with a prefix will be routed to the SIP trunk.
B
Schedule
The schedule is set to Always, meaning that the rule will always be
verified.
C
Status
The status is set On, meaning that the rule is currently active.
D
Direction from
The field is set to local to look for calls that will be originating from
Sipelia Server.
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Value
letter
Value label
Description
E
Direction to
The field is set to TrunkSIPServerA because calls will be routed to the
SIP trunk.
F
Source
The regular expression is set to (.*), meaning that the call can be
made from any SIP extension registered to Sipelia Server. If you
choose to enter a specific SIP extension, make sure that the regular
expression corresponds to the caller's extension.
G
Destination
The prefix ^780 is used. This means that SIP extensions on
Sipelia Server must first dial 780 to be able to communicate with
TrunkSIPServerA. Furthermore, the prefix is followed by the regular
expression (.*) This regular expression creates a regular expression
capturing group for all digits that follow the prefix.
Example: If you want to call extension 1001 on TrunkSIPServerA, you
must dial 7801001. The regular expression (.*) creates a group index
of extensions that begin with 1. As a result, the extension 1001 is
included in this index. If you call 7801002, the extension 1002 will
also be included in this index.
H
New source
The regular expression (.*) is used. This means that the caller's
extension on Sipelia Server remains unchanged.
Example: If the source extension registered on Sipelia Server is 6001,
the dial plan rule will not cause it to change. With this rule, the same
applies to all other registered extensions on Sipelia Server. However,
if you enter a New source value of 7001, and you make a call from
your extension (6001), the recipient sees that the incoming call is
coming from extension 7001, not 6001.
I
New destination
Because the New destination value is linked to the Destination field,
the \1 indicates that the rule must use the value of the first regular
expression group of Destination. In this scenario, the Destination
value is 780(.*), so using \1 will correspond to (.*) which is actually
the destination SIP extension without the prefix.
Result
Once this dial plan rule is imported into Config Tool, if any SIP extension registered to Sipelia Server
dials 7801001, the extension 1001 on SIP Server A will ring.
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Dial plan scenario 2: Reserving a range of SIP extensions for
local calls
In this scenario, dial plan rules are used to define a range of SIP extensions for calls that remain local to
Sipelia Server, while calls made to any other SIP extensions are automatically routed to a SIP trunk.
Scenario
• What do you want to do? You want the SIP extensions 4000 to 4500 reserved for calls that must
remain local to Sipelia Server, while calls made to other SIP extensions are routed to SIP Server A.
• What you need to do before defining the rule? Add a SIP trunk in Config Tool for SIP Server A and
name it TrunkSIPServerA. This trunk name must be unique, so that the dial plan rule that you will
create will not conflict with other rules, and your rule will be applied correctly.
• How can you do it? You need to define two separate rules listed in the right order in your dial plan.
• The first rule will keep the calls local when the destination SIP extensions are between 4000 and
4500.
• The second rule will take calls to any other SIP extensions and route them to TrunkSIPServerA.
IMPORTANT: The rules need to be in the right order in the dial plan to work as described in this
scenario.
Example of the dial plan
The following is an example of a dial plan that accomplishes the above mentioned scenario.
1 Scenario2-OnlyLocalCalls;00000000-0000-0000-0000-000000000006;ON;local;local;(.*);\b0*4([0-4]
[0-9]{2}|500)\b;(.*);(.*);
2 Scenario2RestCallsToMyTrunk;00000000-0000-0000-0000-000000000006;ON;local;TrunkSIPServerA;(.*);(.*);
(.*);(.*);
Rule 1: Only local calls
The first rule listed in the dial plan and shown below tells Sipelia Server to look for source SIP
extensions between 4000 and 4500, and keep them local.
Scenario2-OnlyLocalCalls;00000000-0000-0000-0000-000000000006;ON;
local;local;(.*);\b0*4([0-4][0-9]{2}|500\b;(.*);(.*);
The value labels identified below correspond to the column labels which appear on the Dial plans
page of the Sipelia Plugin role.
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Value
letter
Value label
Description
A
Name
The name indicates that this rule will route local calls.
B
Schedule
The schedule is set to Always, meaning that the rule will always be
verified.
C
Status
The status is set On, meaning that the rule is currently active.
D
Direction from
The field is set to local to look for calls that will be originating from
Sipelia Server.
E
Direction to
The field is set to local because the calls that match the rule will
remain local to Sipelia Server.
F
Source
The regular expression is set to (.*), meaning that the call can be
made from any SIP extension registered to Sipelia Server.
G
Destination
The regular expression is set to \b0*4([0-4][0-9]{2}|500)\b to look
for destination SIP extensions between 4000 and 4500, as described
below:
• \b: SIP extension boundary. Used together with the same \b
element at the end of the expression, this requires the whole SIP
extension to be matched. Starting or ending characters will not be
omitted.
• 0*: Look for any number of zeros before the next character,
which is 4.
• 4: Look for a SIP extension that begins with 4.
• ([0-4]: Look for a single occurrence of digits 0 through 4 after the
first 4.
• [0-9]{2}: Look for 2 following occurrences of digits 0 through 9,
which now covers extension 4000 to 4499.
• |500): Look specifically for 500 to add 4500 in the search pattern.
• \b: Match a SIP extension boundary.
H
New source
The regular expression (.*) is used. This means that your extension
(the source caller) on Sipelia Server remains unchanged.
I
New destination
The regular expression (.*) is used. This means that the called
extension remains unchanged.
Rule 2: Other calls routed to the SIP trunk
The second rule listed in the dial plan and shown below tells Sipelia Server to route to TrunkSIPServerA
any call that is not matching the first rule.
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Scenario2-RestCallsToMyTrunk;00000000-0000-0000-0000-000000000006;ON;
local;TrunkSIPServerA;(.*);(.*);(.*);(.*);
Value
letter
Value label
Description
A
Name
The name indicates that this rule will route any other call to
TrunkSIPServerA.
B
Schedule
The schedule is set to Always, meaning that the rule will always be
verified.
C
Status
The status is set On, meaning that the rule is currently active.
D
Direction from
The field is set to local to look for calls that will be originating from
Sipelia Server.
E
Direction to
The field is set to TrunkSIPServerA to route the calls to that SIP trunk.
F
Source
The regular expression is set to (.*), meaning that the call can be
made from any SIP extension registered to Sipelia Server.
G
Destination
The regular expression is set to (.*), meaning that the call can be
made to reach any SIP extension.
H
New source
The regular expression (.*) is used. This means that your extension
(the source caller) on Sipelia Server remains unchanged.
I
New destination
The regular expression (.*) is used. This means that the called
extension remains unchanged.
Result
Once the dial plan is imported into Config Tool, only the highest priority rule will apply. This dial plan
results in the following:
1 If any SIP extension dials a number between 4000 and 4500, the call will remain local to Sipelia
Server.
2 If any SIP extension dials any other number, the call will automatically be routed to SIP Server A.
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Dial plan scenario 3: Reserving a range of SIP extensions for
calls to a SIP trunk
In this scenario, dial plan rules are used to define a range of SIP extensions for calls that are
automatically routed to a SIP trunk.
Scenario
• What do you want to do? You want to reserve SIP extensions 4000 to 4500 for calls that must
automatically be routed to SIP Server A.
• What you need to do before defining the rule? Add a SIP trunk in Config Tool for SIP Server A and
name it TrunkSIPServerA. This trunk name must be unique, so that the dial plan rule that you will
create will not conflict with other rules, and your rule will be applied correctly.
• How can you do it? You need to define two separate rules listed in the right order in your dial plan.
• The first rule will take any call made to a SIP extension that is between 4000 and 4500 and
automatically route it to TrunkSIPServerA.
• The second rule will take calls to any other SIP extension and will keep it local to Sipelia Server.
IMPORTANT: The rules need to be in the right order in the dial plan to work as described in this
scenario.
Example of the dial plan
The following is an example of a dial plan that accomplishes the above mentioned scenario.
1 Scenario3OnlyCallsToMyTrunk;00000000-0000-0000-0000-000000000006;ON;local;TrunkSIPServerA;(.*);
\b0*4([0-4][0-9]{2}|500)\b;(.*);(.*);
2 Scenario3-RestAreLocalCalls;00000000-0000-0000-0000-000000000006;ON;local;local;(.*);(.*);(.*);
(.*);
Rule 1: Calls to the SIP trunk only
The first rule listed in the dial plan and shown below tells Sipelia Server to look for destination SIP
extensions between 4000 and 4500, and forward them to the SIP trunk.
Scenario3-OnlyCallsToMyTrunk;00000000-0000-0000-0000-000000000006;ON;
local;TrunkSIPServerA;(.*);\b0*4([0-4][0-9]{2}|500\b;(.*);(.*);
The value labels identified below correspond to the column labels which appear on the Dial plans
page of the Sipelia Plugin role.
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Value
letter
Value label
Description
A
Name
The name indicates that this rule will route to the SIP trunk calls
made with specific SIP extensions.
B
Schedule
The schedule is set to Always, meaning that the rule will always be
verified.
C
Status
The status is set On, meaning that the rule is currently active.
D
Direction from
The field is set to local to look for calls that will be originating from
Sipelia Server.
E
Direction to
The field is set to TrunkSIPServerA to route the calls to that SIP trunk.
F
Source
The regular expression is set to (.*), meaning that the call can be
made from any SIP extension registered to Sipelia Server.
G
Destination
The regular expression is set to \b0*4([0-4][0-9]{2}|500)\b, meaning
the Sipelia Server must to look for destination SIP extensions
between 4000 and 4500, as described below:
• \b: SIP extension boundary. Used together with the same \b
element at the end of the expression, this requires the whole SIP
extension to be matched. Starting or ending characters will not be
omitted.
• 0*: Look for any number of zeros before the next character,
which is 4.
• 4: Look for a SIP extension that begins with 4.
• ([0-4]: Look for a single occurrence of digits 0 through 4 after the
first 4.
• [0-9]{2}: Look for 2 following occurrences of digits 0 through 9,
which now covers extension 4000 to 4499.
• |500): Look specifically for 500 to add 4500 in the search pattern.
• \b: Match a SIP extension boundary.
H
New source
The regular expression (.*) is used. This means that your extension
(the source caller) on Sipelia Server remains unchanged.
I
New destination
The regular expression (.*) is used. This means that the called
extension remains unchanged.
Rule 2: Other calls routed locally
The second rule listed in the dial plan and shown below tells Sipelia Server to route to TrunkSIPServerA
any other call that does not match the first rule.
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Scenario3-RestAreLocalCalls;00000000-0000-0000-0000-000000000006;ON;
local;local;(.*);(.*);(.*);(.*);
Value
letter
Value label
Description
A
Name
The name indicates that this rule will keep local any other call.
B
Schedule
The schedule is set to Always, meaning that the rule will always be
verified.
C
Status
The status is set On, meaning that the rule is currently active.
D
Direction from
The field is set to local to look for calls that will be originating from
Sipelia Server.
E
Direction to
The field is set to local to keep the calls local to Sipelia Server.
F
Source
The regular expression is set to (.*), meaning that the call can be
made from any SIP extension registered to Sipelia Server.
G
Destination
The regular expression is set to (.*), meaning that the call can be
made to reach any SIP extension.
H
New source
The regular expression (.*) is used. This means that your extension
(the source caller) on Sipelia Server remains unchanged.
I
New destination
The regular expression (.*) is used. This means that the called
extension remains unchanged.
Result
Once the dial plan is imported into Config Tool, only the highest priority rule will be apply. This dial plan
results in the following:
1 If any SIP extension dials a number between 4000 and 4500, the call will be routed to
TrunkSIPServerA.
2 If any SIP extension dials any other number, the call will remain local to Sipelia Server.
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Dial plan scenario 4: Replacing source SIP extensions
In this scenario, a dial plan rule is used to replace the source SIP extensions with a unique SIP extension
that will be displayed to the recipients. This rule can be used to ensure the privacy of the callers.
Scenario
• What do you want to do? Display a unique source SIP extension for any call that is made.
• What you need to do before defining the rule? Select the SIP extension that will always be used as
the source. In this scenario, 123456 is the selected SIP extension.
• How can you do it? You need to create a rule that takes any call from Sipelia Server and replace the
source SIP extension with 123456.
Example of the dial plan rule
The following is an example of a dial plan rule that accomplishes the above mentioned scenario.
Scenario4-ChangeSource;00000000-0000-0000-0000-000000000006;ON;
local;local;(.*);(.*);123456;(.*);
The value labels identified below correspond to the column labels which appear on the Dial plans
page of the Sipelia Plugin role.
Value
letter
Value label
Description
A
Name
The name of the rule indicates that it will replace the source SIP
extension.
B
Schedule
The schedule is set to Always, meaning that the rule will always be
verified.
C
Status
The status is set On, meaning that the rule is currently active.
D
Direction from
The field is set to local to look for calls that will be originating from
Sipelia Server.
E
Direction to
The field is set to local to keep the calls local to Sipelia Server.
F
Source
The regular expression is set to (.*), meaning that the call can be
made from any SIP extension registered to Sipelia Server.
G
Destination
The regular expression is set to (.*), meaning that the call can be
made to reach any SIP extension.
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Value
letter
Value label
Description
H
New source
The regular expression is set to 123456 to change the SIP extension
with this constant value.
I
New destination
The regular expression (.*) is used. This means that the called
extension remains unchanged.
Result
Once this dial plan rule is imported into Config Tool, if any SIP extension registered to Sipelia Server
dials any other SIP extension, the source extension 123456 will be displayed to the recipients.
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Dial plan scenario 5: Removing prefix on source SIP
extensions from a SIP trunk
In this scenario, a dial plan rule is used to remove the prefix used in source SIP extensions for calls
received from a SIP trunk, so that it is not displayed to the recipients. This scenario can be combined
with scenario 1 to have the source extensions displayed with no prefix when received by the recipients
registered to SIP Server A.
Scenario
• What do you want to do? Remove the prefix included in source SIP extensions when calls are
received from a SIP trunk.
• What you need to do before defining the rule? Add a SIP trunk in Config Tool for SIP Server A and
name it TrunkSIPServerA. This trunk name must be unique, so that the dial plan rule that you will
create will not conflict with other rules, and your rule will be applied correctly. You also need to
know the prefix used by the SIP trunk for its extensions.
• How can you do it? Create a rule that takes any call from a SIP trunk and remove the prefix used in
the source SIP extension.
Example of the dial plan rule
The following is an example of a dial plan rule that accomplishes the above mentioned scenario.
Scenario5-RemoveSourceHeaderFromTrunk;00000000-0000-0000-0000-000000000006;
ON;TrunkSIPServerA;local;^551(.*);(.*);\1;(.*);
The value labels identified below correspond to the column labels which appear on the Dial plans
page of the Sipelia Plugin role.
Value
letter
Value label
Description
A
Name
The name of the rule indicates that it will remove the prefix from
source SIP extensions received from the SIP trunk.
B
Schedule
The schedule is set to Always, meaning that the rule will always be
verified.
C
Status
The status is set On, meaning that the rule is currently active.
D
Direction from
The field is set to TrunkSIPServerA to look for calls that will be
originating from the SIP trunk.
E
Direction to
The field is set to local to route the calls to Sipelia Server.
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Value
letter
Value label
Description
F
Source
The regular expression is set to ^551(.*) to look for any source SIP
extension that begins with prefix 551.
G
Destination
The regular expression is set to (.*), meaning that the call can be
made to reach any SIP extension.
H
New source
I
New destination
The regular expression is set to \1, which indicates that the rule
will use the value of the first regular expression group, which was
defined in the Source value. This means that the prefix will be
discarded and the source SIP extension becomes (.*).
The regular expression (.*) is used. This means that the called
extension remains unchanged.
Result
Once this dial plan rule is imported into Config Tool, any source SIP extension with prefix 551 received
from TrunkSIPServerA will be displayed without the prefix to the recipients. For example, if a call
received from the SIP trunk is made from extension 5513005, the recipient registered to Sipelia Server
will only see 3005.
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Dial plan scenario 6: Forwarding calls to another SIP
extension on schedule
In this scenario, dial plan rules are used to define a range of SIP extensions for calls that must always
reach the recipients, while other SIP extensions are automatically forwarded to a new destination. This
can be useful to route calls from SIP intercoms to a call center during the night for example.
Scenario
• What do you want to do? During off-hours for example, calls made to SIP extensions between 4000
and 4500 must be routed normally to the requested recipients, while other calls must be forwarded
to extension 1001, which can be the extension of a SIP endpoint (or SIP client) in a security office for
example.
• What you need to do before defining the rule? Define the range of SIP extensions that must always
be routed and the specific SIP extension to which other calls will be forwarded. You also need to
have a schedule defined in Security Center to be used in the dial plan rule.
• How can you do it? You need to define two separate rules listed in the right order in your dial plan:
• The first rule looks for any call made to reach SIP extensions between 4000 to 4500, and route
them normally.
• The second rule will take calls to any other SIP extension and forward them to extension 1001.
IMPORTANT: The rules need to be in the right order in the dial plan to work as described in this
scenario.
Example of the dial plan
The following is an example of a dial plan that accomplishes the above mentioned scenario.
1 Scenario6-NightRoutingNormal;DB3ABC7D-FD1B-4A92-A392-4408404F7D7B;ON;local;local;(.*);
\b0*4([0-4][0-9]{2}|500)\b;(.*);(.*);
2 Scenario6-NightRoutingSpecial;DB3ABC7D-FD1B-4A92-A392-4408404F7D7B;ON;local;local;(.*);(.*);
(.*);1001;
Rule 1: Normal night routing
The first rule listed in the dial plan and shown below tells Sipelia Server to look for destination SIP
extensions between 4000 and 4500, and route them normally.
Scenario6-NightRoutingNormal;DB3ABC7D-FD1B-4A92-A392-4408404F7D7B;ON;
local;local;(.*);\b0*4([0-4][0-9]{2}|500)\b;(.*);(.*);
The value labels identified below correspond to the column labels which appear on the Dial plans
page of the Sipelia Plugin role.
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SIP trunks and dial plans
Value
letter
Value label
Description
A
Name
The name of the rule indicates that the calls are routed normally.
B
Schedule
The schedule is set to Off-hours, meaning that the rule will be
applied only during that time.
C
Status
The status is set On, meaning that the rule is currently active.
D
Direction from
The field is set to local to look for calls that will be originating from
Sipelia Server.
E
Direction to
The field is set to local because the calls will remain local to Sipelia
Server when the rule is applied.
F
Source
The regular expression is set to (.*), meaning that the call can be
made from any SIP extension.
G
Destination
The regular expression is set to \b0*4([0-4][0-9]{2}|500)\b, meaning
that Sipelia Server must look for called SIP extensions between 4000
and 4500, as described below:
• \b: SIP extension boundary. Used together with the same \b
element at the end of the expression, this requires the whole SIP
extension to be matched. Starting or ending characters will not be
omitted.
• 0*: Look for any number of zeros before the next character,
which is 4.
• 4: Look for a SIP extension that begins with 4.
• ([0-4]: Look for a single occurrence of digits 0 through 4 after the
first 4.
• [0-9]{2}: Look for 2 following occurrences of digits 0 through 9,
which now covers extension 4000 to 4499.
• |500): Look specifically for 500 to add 4500 in the search pattern.
• \b: Match a SIP extension boundary.
H
New source
The regular expression (.*) is used. This means that your extension
(the source caller) on Sipelia Server remains unchanged.
I
New destination
The regular expression (.*) is used. This means that the called
extension remains unchanged.
Rule 2: Special night routing
The second rule listed in the dial plan and shown below tells Sipelia Server to look for any call that does
not match the first rule, and forward it to extension 1001.
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SIP trunks and dial plans
Scenario6-NightRoutingSpecial;DB3ABC7D-FD1B-4A92-A392-4408404F7D7B;
ON;local;local;(.*);(.*);(.*);1001;
The value labels identified below correspond to the column labels which appear on the Dial plans
page of the Sipelia Plugin role.
Value
letter
Value label
Description
A
Name
The name of the rule indicates that the calls are following a special
route.
B
Schedule
The schedule is set to Off-hours, meaning that the rule will be
applied only during time.
C
Status
The status is set On, meaning that the rule is currently active.
D
Direction from
The field is set to local to look for calls that will be originating from
Sipelia Server.
E
Direction to
The field is set to local because the calls will remain local to Sipelia
Server when the rule is applied.
F
Source
The regular expression is set to (.*), meaning that the call can be
made from any SIP extension.
G
Destination
The regular expression is set to (.*), meaning that the call can be
made to reach any SIP extension.
H
New source
The regular expression (.*) is used. This means that your extension
(the source caller) on Sipelia Server remains unchanged.
I
New destination
The regular expression 1001 is used. This means that the call will
always be forwarded to SIP extension 1001.
Result
Once the dial plan is imported into Config Tool, only the highest priority rule will apply. This dial plan
results in the following:
1 When the schedule is satisfied, if any SIP extension dials a number between 4000 and 4500, the call
will remain local to Sipelia Server and will be routed normally to the requested recipient.
2 When the schedule is satisfied, if any SIP extension dials any other number, the call will
automatically be forwarded to SIP extension 1001.
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5
Troubleshooting
This section includes the following topics:
•
"Troubleshooting: Unable to establish communication with server" on page 69
•
"Troubleshooting: Message broker connection failed" on page 70
•
"Troubleshooting: Cannot add SIP intercom devices" on page 71
•
"Troubleshooting: Cannot see the Sipelia icon in the notification tray" on page 72
•
"Troubleshooting: Security Desk cannot connect to Sipelia Server" on page 73
•
"Troubleshooting: Cannot register to Sipelia Server from Security Desk" on page 74
•
"Troubleshooting: Cannot make calls between two SIP endpoints" on page 75
•
"Troubleshooting: No video displayed during calls" on page 76
•
"Troubleshooting: Audio and video not being recorded" on page 77
•
"Troubleshooting: Users cannot view recorded video" on page 78
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Troubleshooting
Troubleshooting: Unable to establish communication with
server
When adding a SIP intercom in Config Tool, the error message Unable to establish communication
with the server is displayed.
What you should know
This error typically occurs when there is a connection issue with the configuration service between
Config Tool and Sipelia Server.
To troubleshoot this issue, try the following:
1
2
3
4
Log on to Security Center with Config Tool, and then open the Plugins task.
Select the Sipelia Plugin role, and then click General.
Make sure that Configuration service port is set to the proper value.
Make sure that this port is not being blocked anywhere on the network.
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Troubleshooting
Troubleshooting: Message broker connection failed
The Sipelia Plugin role displays the error message Message broker connection failed to: and
connection to Sipelia Server is not possible from Security Desk.
What you should know
This error typically occurs when the version of RabbitMQ already installed on your server is more
recent than the version required by Sipelia. Changes made to subsequent RabbitMQ versions prevent
the use of the more recent versions. It is recommended to install the version that comes with the
Sipelia installation package.
To establish a connection with the message broker, try the following:
1
2
3
4
Restart the Genetec Server.
If the problem is still present, continue with the next steps.
Uninstall RabbitMQ Server from your computer.
Remove all the files and folders located in C:\Users\<your_user_name>\AppData\Roaming
\RabbitMQ.
NOTE: <your_user_name> corresponds to the user account that was used when RabbitMQ was
originally installed.
5 If you get the message You need permission to perform this action, delete all the files first, then
delete the folders.
6 Install the RabbitMQ version that comes with the Sipelia installation package.
NOTE: The RabbitMQ installation package is normally found in the ISSetupPrerequisites folder.
7 Select all components, then click Next.
8 Select the destination folder, then click Install.
9 When the installation has completed, click Finish.
10 Restart the Sipelia Plugin role.
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Troubleshooting
Troubleshooting: Cannot add SIP intercom devices
When adding a SIP intercom in Config Tool, the error message The number of Sipelia licenses
(standard or advanced) has been exceeded. is displayed.
What you should know
This error typically occurs when you do not have enough standard and/or advanced licenses in your
system to support the number of SIP intercom devices that you want to add.
To be able to add new SIP intercom devices, do one of the following:
• Remove one or more SIP intercom devices from your system.
• Increase the number of licenses.
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Troubleshooting
Troubleshooting: Cannot see the Sipelia icon in the
notification tray
A user cannot see the Sipelia icon in the notification tray of Security Desk.
What you should know
This issue typically occurs when the Sipelia Client has not been installed properly.
To troubleshoot this issue, try the following:
1 Restart Security Desk.
2 If the problem is still present, restart the computer on which Security Desk is running.
3 If you are still unable to see the icon in the notification tray, try to uninstall and re-install the Sipelia
Client on the computer.
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Troubleshooting
Troubleshooting: Security Desk cannot connect to Sipelia
Server
If Security Desk cannot connect to the Sipelia Server, verify that the IP address and port properties are
properly set.
What you should know
Connection issues typically occur when IP addresses or ports are not configured properly, or when the
network is blocking packets from being exchanged between two endpoints.
To troubleshoot connection issues to the Sipelia Server, try the following:
1 If the computer on which Security Desk is running provides multiple network interfaces (cards),
verify that the network interface used for Sipelia is configured with the highest priority.
2 Verify that the RabbitMQ Server is installed and is running on both the client (Security Desk) and
server (Sipelia Server).
3 Log on to Security Center with Config Tool, and then open the Plugins task.
4 Select the Sipelia Plugin role, and verify that the role is up and running.
5 Click General, and make sure that the address and port properties are set to the proper values.
6 Click Servers, and make sure that SIP port is set to the proper value.
7 Make sure that the ports are not being blocked anywhere on the network.
8 Open the Network task, and then select the server that is hosting the Sipelia Plugin role.
9 Click Properties, and make sure that the first IP address shown in Private addresses, that is, the IP
address listed at the top, is the one that you want to be used by the server.
Sipelia Server automatically uses the first IP address shown in the list.
10 Log on to Security Center with Security Desk.
11 Click Options > Sipelia > Advanced, and make sure that UDP port range is set properly.
12 On the Sipelia Server, browse to folder C:\ProgramData\Genetec Sipelia 2.0\SipServer and open
SipServer.config.
13 Make sure that MinimumPortRange and MaximumPortRange values are set properly in the file.
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Troubleshooting
Troubleshooting: Cannot register to Sipelia Server from
Security Desk
If users cannot register to Sipelia Server from Security Desk, you should verify that the VoIP properties
are properly set for those users.
What you should know
Registration issues typically occur when Security Desk can connect to the Sipelia Server but SIP
extensions or passwords are not configured properly.
To troubleshoot this issue, try the following:
1 Log on to Security Center with Config Tool.
2 Open the Security task, and then select the user entity involved in the connection issue.
3 In the VoIP page, make sure that the SIP extension and Password properties have been configured
with the proper values.
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Troubleshooting
Troubleshooting: Cannot make calls between two SIP
endpoints
If users cannot make or receive calls with Security Desk, the messages might be blocked somewhere on
the network.
What you should know
When users are registered to Sipelia Server but cannot make or receive calls, this typically occurs when
the network is blocking packets from being exchanged between two SIP endpoints.
To troubleshoot this issue, try the following:
1 Log on to Security Center with Security Desk.
2 Using a network protocol analyzer (Wireshark for example), try to make a call and verify that SIP
messages are exchanged between the two SIP endpoints.
3 If SIP packets are being exchanged, verify that SDP and RTP packets are also exchanged.
4 Make sure that the IP addresses and port numbers used in the SIP and SDP packets are correct.
5 If you can see SIP, SDP, and RTP packets exchanged on the network between the two SIP endpoints
but users are still not able to make or receive calls, try to restart the Genetec Server.
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Troubleshooting
Troubleshooting: No video displayed during calls
If you cannot see video during calls, you might have video codec options that are not configured
properly.
To troubleshoot this issue, try the following:
1 Log on to Security Center with Security Desk.
2 Click Options > Sipelia > Advanced.
3 Make sure that the following is set correctly:
• Video codecs: The video codecs that are supported by Security Desk for video communication.
By default, the H.264 and H.263 codecs are turned on, and should suffice for most cases. As
a result, it is recommended to keep the default settings, and to be aware that changing video
codecs can disrupt the video that is streamed during video calls. To be able to view video during
a SIP video call, the SIP clients that are involved in a call must all support at least one common
video codec. For example, if SIP client A only supports the H.264 codec and SIP client B only
supports H.263, no video is streamed during a call session between the two SIP clients.
• UDP port range: The port range for the User Diagram Protocol (UDP). The UDP ports are used
by the different SIP clients to send and receive communication data. The default range is from
20000 to 20500. It is recommended to keep the default settings, and to change them only if
Sipelia logs any port-related issues about making or receiving calls with Security Desk.
NOTE: The highest matching preference will be chosen. If both endpoints support H.264 and H.263,
the connection will be established using H.264.
4 If you still have an issue, using a network protocol analyzer (Wireshark for example), verify that SDP
and RTP packets are exchanged between the two SIP endpoints.
5 If you can see SDP and RTP packets exchanged on the network but the there is still no video
displayed during calls, try to restart the Genetec Server.
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Troubleshooting
Troubleshooting: Audio and video not being recorded
If you can see video during calls but you have no recording in the database, you might have recording
options that are not configured properly.
To make sure that audio and video are recorded during call sessions:
1
2
3
4
5
Make sure that your license supports recording of call sessions.
Log on to Security Center with Config Tool, and then open the Plugins task.
Select the Sipelia Plugin role, and then click Recording.
Make sure that User recording and Device recording are enabled.
If you have no recording associated with specific users, open the Security task, and then select the
user entity involved in the issue.
6 In the VoIP page, make sure that the Record audio and video property is turned on or inherits the
default value from the Sipelia Plugin role.
7 If you still have problems to record call sessions, try to restart the Genetec Server.
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Troubleshooting
Troubleshooting: Users cannot view recorded video
If a user cannot see the video that was recorded during previous call sessions, you might have user
privileges that are not set properly.
What you should know
This issue typically occurs when the user does not have the privilege to view recorded video in Security
Center.
To enable users to view recorded video:
1 Log on to Security Center with Config Tool.
2 Open the Security task, and then select the user entity involved in the issue.
3 In the Privileges page, make sure that the View playback privilege is allowed for the user.
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Appendices
Additional resources
This section includes the following topics:
•
"Common VoIP terms" on page 80
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A
Common VoIP terms
This section includes the following topics:
•
"Common VoIP terms" on page 81
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Common VoIP terms
Common VoIP terms
A VoIP (voice over Internet Protocol) environment needs many interacting components to work
properly. Below is a list of common industry terms that are used throughout the various Sipelia help
guides.
• dual-tone multifrequency (DTMF): Dual-tone multifrequency is the standard for the audible signal
that is sent to the phone company once a key is pressed on a telephone keypad. There is an audible
tone to represent each digit on a keypad.
• IP phone: An Internet Protocol (IP) phone is a device that is used to make and receive calls over the
Internet. An IP phone can use any of the existing communication standards or protocols, such as
SIP, to transmit calls across a network. Although an IP phone can look like a traditional phone, an
IP phone is not connected to a phone-line jack found in typical POTS installations; an IP phone is
connected to a router or RJ-45 Ethernet connector.
• plain old telephone system (POTS): The plain old telephone system (POTS) is the basic form
of telephone service that is used by most homes and businesses worldwide. Apart from being
a different technology, what separates a non-POTS service from a POTS service is speed and
bandwidth. POTS is also known as the public switched telephone network (PSTN).
• private branch exchange (PBX): A private branch exchange (PBX) is a private phone network that
is used within a company. A PBX is a switching station that is used to connect many internal phone
extensions to one outside line, thereby making it more efficient and cost effective for companies to
adopt a phone system. In a company that uses a PBX network, incoming calls are redirected by the
PBX to one or more internal extensions within the same enterprise.
• SIP trunking: SIP trunking is the process of using VoIP technology to connect existing PBX systems
to other PBX systems. SIP trunking replaces traditional phone trunks with an IP network and
consolidates voice, data, and video into a single trunk (or line). SIP trunking ensures a more reliable
communication service and cost reductions.
• softphone: A softphone is a software for managing inbound and outbound calls over a network,
using your computer rather than a phone. Softphones are designed to simulate the functions found
on traditional phones. Also known as SIP client.
• Voice over Internet Protocol (VoIP): Voice over Internet Protocol (VoIP) Is the technology for
routing two-way voice and video communications over the web or other IP networks.
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Glossary
A B C D E F G H I J K L M N O P Q R S T U V W X Y Z
C
call dialog box
The call dialog box is a Sipelia-related dialog box that appears from the notification
tray within Security Desk once an incoming call comes in from a SIP endpoint. In the
call dialog box, users can manage their calls and their list of contacts and favorites,
and set their availability status.
Call report
The Call report task is a type of investigation task that allows users to review call
sessions and generate reports. Among the possible actions that can be performed
in this task, users can view the call logs for all sessions, watch playback video of
all recorded call sessions, and see the bookmarks that have been added to video
sequences of associated cameras.
call session
A call session is the sequence of events or activities that occur from the time a SIP
call is initiated to the time that it ends, including all call transfers. For example, if a
call is transferred twice, the sequence of events that occur in both those transfers
are all part of the same call session. In a Security Center system equipped with the
Sipelia module, call sessions can be reviewed and exported through the Call report
task within Security Desk.
Config Tool
Config Tool is a Security Center administrative application used to manage all
Security Center users, and configure all Security Center entities such as areas,
cameras, doors, schedules, cardholders, Patroller/LPR units, and hardware devices.
conversation window
The conversation window is a Sipelia-related window that opens within Security
Desk once a SIP call has been accepted by either the caller or the recipient of a call.
From the conversation window, Security Center users can manage conversations,
forward calls, and view associated video when available.
D
dial plan
A dial plan is a collection of rules that defines how calls are routed locally or
between two SIP trunks. Dial plans ensure that calls are routed and rerouted
correctly, and they also allow administrators to block calls to certain geographic
locations or ensure the privacy of the callers.
E
entity
Entities are the basic building blocks of Security Center. Everything that requires
configuration is represented by an entity. An entity can represent a physical device,
such as a camera or a door, or an abstract concept, such as an alarm, a schedule, a
user, a role, a plugin, or an add-on.
G
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Glossary
Genetec Server
Genetec Server is the Windows service that is at the core of Security Center
architecture, and that must be installed on every computer that is part of the
Security Center's pool of servers. Every such server is a generic computing resource
capable of taking on any role (set of functions) you assign to it.
N
notification tray
The notification tray contains icons that allow quick access to certain system
features, and also displays indicators for system events and status information. The
notification tray display settings are saved as part of your user profile and apply to
both Security Desk and Config Tool.
R
regular expression
A regular expression is a sequence of symbols used by a regular expression
engine to identify all the strings of characters that match a specific search pattern
without having to list all the possible discrete values that must be returned. The
Microsoft's .NET Framework Regular Expression engine is the engine used in Sipelia.
ring group
A ring group is a group of SIP entities that has its own unique phone extension. All
entities (or members) within a ring group are part of a call list, and all members get
called when the ring group extension is called. The members of a ring group can
either be called all at once, or successively at a set interval. The call stops ringing
once any one of the members within a call list answers the call.
role
A role is a software module that performs a specific job within Security Center.
Roles must be assigned to one or more servers for their execution.
S
Security Center
Security Center is the unified security platform that seamlessly blends Genetec's IP
security and safety systems within a single innovative solution. The systems unified
under Security Center include Genetec's Omnicast IP video surveillance system,
Synergis IP access control system, and AutoVu IP license plate recognition (LPR)
system.
Security Desk
Security Desk is the unified user interface of Security Center. It provides consistent
operator flow across all of the Security Center’s main systems, Omnicast, Synergis,
and AutoVu. Security Desk’s unique task-based design lets operators efficiently
control and monitor multiple security and public safety applications.
Session Initiation
Protocol
The Session Initiation Protocol (SIP) is a signalling standard that is used to control
the exchange of data between two or more parties in multimedia communications.
Sipelia, the Security Center core module that allows users to make, receive, and
manage voice and video calls over the web, is based on the SIP standard.
SIP client
A SIP client is a program with softphone functionality that users can install on their
computers or mobile devices to make and receive voice or video calls from other
SIP clients. A SIP client requires a SIP account, and typically includes a user interface
from which users can manage calls and also view call-related video streams, if such
a feature is supported. Examples of SIP clients are SIP phones, softphones, and SIP
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Glossary
intercoms. Once Sipelia Client is installed and configured, Security Desk is also an
example of a SIP client.
Sipelia
Sipelia is a core module of Security Center that allows Security Center users to
make, receive, and manage SIP-based voice and video calls over a network. Running
on the open source Session Initiation Protocol (SIP), Sipelia also integrates existing
video and access control platforms with intercom systems, and allows users to log
call activities.
Sipelia Client
Sipelia Client is the softphone component of Sipelia. As a result, it installs the
various user interface features of the Sipelia module, such as the call dialog box
and conversation window. Sipelia Client must be installed on every Security Desk
workstation that is running Sipelia, thus turning Security Desk into a SIP client (or
softphone).
Sipelia Server
Sipelia Server is the SIP server component of Sipelia. It receives and administers
information about the different SIP endpoints, and essentially facilitates the
communication between two or more endpoints that are communicating in a SIP
environment. Sipelia Server also collates and stores important data, such as contact
list information, SIP server settings, and call session recordings. Sipelia Server must
be run by a Security Center Plugin role, and therefore, must be installed on every
Security Center server on which you intend to host the Plugin role.
SIP endpoint
A SIP endpoint is the device or system that is at each end of a SIP call session.
Examples of endpoints are hard-wired phones, voice mail systems, and intercoms.
A SIP client such as a softphone, is another example of an endpoint. Once Sipelia
Client is installed and configured, Security Desk is considered both an endpoint and
a SIP client.
SIP entity
A SIP entity is a Security Center entity that has SIP-related capabilities. In Security
Center, examples of SIP entities are users, ring groups, and SIP devices such as SIP
intercoms.
SIP extension
A SIP extension is a numeric value assigned to a SIP device so that the device can
make and receive SIP calls. Typically, a SIP extension is also used to register the
SIP device (to which it is assigned) to a SIP server. To be able to communicate with
other SIP endpoints, every SIP entity (user, ring group, or intercom) in Security
Center must have a unique SIP extension assigned to it.
SIP intercom
A SIP intercom is an intelligent SIP endpoint that provides two-way phone
connectivity in a SIP environment. In Security Center, a SIP intercom is one of the
established SIP entities, and it is the only SIP entity that is an actual device. The
other SIP entities are Security Center users and ring groups.
SIP trunk
A SIP trunk is a SIP server that allows users to connect their existing SIP servers to
other servers, thus extending their VoIP capabilities and allowing them to migrate
their old PBX systems to a unified VoIP system. With an integrated dial plan, SIP
trunks make it possible for SIP extensions that reside on different SIP servers to
communicate with one another.
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Where to find product information
You can find our product documentation in the following locations:
• Installation package: The documentation is available in the Documentation folder of the installation
package. Some of the documents also have a direct download link to the latest version of the
document.
• Genetec Technical Assistance Portal (GTAP): The latest version of the documentation is available
from the GTAP Documents page. Note, you’ll need a username and password to log on to GTAP.
• Help: Security Center client and web-based applications include help, which explain how the
product works and provide instructions on how to use the product features. Patroller and the Sharp
Portal also include context-sensitive help for each screen. To access the help, click Help, press F1, or
tap the ? (question mark) in the different client applications.
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Technical support
Genetec Technical Assistance Center (GTAC) is committed to providing its worldwide clientele with
the best technical support services available. As a Genetec customer, you have access to the Genetec
Technical Assistance Portal (GTAP), where you can find information and search for answers to your
product questions.
• Genetec Technical Assistance Portal (GTAP): GTAP is a support website that provides in-depth
support information, such as FAQs, knowledge base articles, user guides, supported device lists,
training videos, product tools, and much more.
Prior to contacting GTAC or opening a support case, it is important to look at this website for
potential fixes, workarounds, or known issues. You can log in to GTAP or sign up at https://
gtap.genetec.com.
• Genetec Technical Assistance Center (GTAC): If you cannot find your answers on GTAP, you can
open a support case online at https://gtap.genetec.com. For GTAC's contact information in your
region see the Contact page at https://gtap.genetec.com.
NOTE: Before contacting GTAC, please have your System ID (available from the About button in
your client application) and your SMA contract number (if applicable) ready.
• Licensing:
• For license activations or resets, please contact GTAC at https://gtap.genetec.com.
• For issues with license content or part numbers, or concerns about an order, please contact
Genetec Customer Service at customerservice@genetec.com, or call 1-866-684-8006 (option #3).
• If you require a demo license or have questions regarding pricing, please contact Genetec Sales
at sales@genetec.com, or call 1-866-684-8006 (option #2).
Additional resources
If you require additional resources other than the Genetec Technical Assistance Center, the following is
available to you:
• GTAP Forum: The Forum is an easy to use message board that allows clients and Genetec staff to
communicate with each other and discuss a variety of topics, ranging from technical questions to
technology tips. You can log in or sign up at https://gtapforum.genetec.com.
• Technical training: In a professional classroom environment or from the convenience of your own
office, our qualified trainers can guide you through system design, installation, operation, and
troubleshooting. Technical training services are offered for all products and for customers with
a varied level of technical experience, and can be customized to meet your specific needs and
objectives. For more information, go to http://www.genetec.com/Services.
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