Administrationshandbuch Mitel TA7102i

TA7102i, Software Configuration Guide
Aastra TA7102i, Software Configuration Guide
Dgw v2.0 Application TA7102i, Software Configuration Guide

Document ID: 154/1531 ANF 901 Uen A 2014-01-30
© Copyright Aastra Technologies Limited, 2014.
Disclaimer
No part of this document may be reproduced in any form without the written permission of the
copyright owner.
The contents of this document are subject to revision without notice due to continued progress in
methodology, design and manufacturing. Aastra shall have no liability for any error or damage of any
kind resulting from the use of this document.
Trademarks
Microsoft and Windows are registered trademarks of Microsoft Corporation.
Adobe and Acrobat are registered trademarks of Adobe Systems Incorporated.
All other trademarks and registered trademarks are the property of their respective owners.
Third-Party Software Copyright Information
The Dgw v2.0 Application firmware aggregates some third-party software modules (open source and
commercial) that are distributed to you in accordance with their respective licenses. Refer to the Third
Party Software Copyright Information addendum, which lists the third-party software modules along
with any copyright and license information.
P R E F A C E
P About this Manual
Document Objectives
The Dgw v2.0 application Software Configuration Guide provides technical information on how to configure
and operate the application for your Aastra unit.
Use the Dgw v2.0 application Software Configuration Guide in conjunction with the appropriate publications
listed in “Related Documentation” on page iii.
Any reference to gateway unit 4102S does relate to terminal adapter TA7102i.
Intended Audience
This Software Configuration Guide is intended for the following users:

 System administrators who are responsible for installing and configuring networking equipment
and who are familiar with the Aastra unit.
 System administrators with a basic networking background and experience, but who might not
be familiar with the Aastra unit.
 Operators.
 Installers.
 Maintenance technicians.
Related Documentation
In addition to this manual, the Aastra unit document set includes the following:
 Model-Specific Hardware Installation Guide
Describes how to install the hardware of your specific Aastra unit.
This booklet allows you to quickly setup and work with the Aastra unit.
Lists all the parameters, tables, and commands available in the Dgw v2.0 application.
Lists and describes all syslog messages and notification messages that the Dgw v2.0 application
may send.
 Third Party Software Copyright Information (please contact your Aastra representative for
detailed information if needed.
This document lists the third-party software modules used in the Aastra unit along with any
copyright and license information.
Be sure to read any readme files, technical bulletins, or additional release notes for important information.
Dgw v2.0 Application iii

Preface - About this Manual Document Structure
Document Structure
The Dgw v2.0 application Software Configuration Guide contains the following information.
Table 1: Software Configuration Guide Chapter/Appendices
Title Summary
“Chapter 1 - System Overview” on page 1 Provides an overview of the Dgw v2.0 application as
well as the units that support it.
“Chapter 2 - Command Line Interface (CLI)” on Describes how to access the CLI environment in order
page 11 to perform configuration tasks.
“Chapter 3 - Web Interface Configuration” on Describes how to access the embedded web server of
page 33 the Aastra unit.
System Parameters
“Chapter 4 - Services” on page 43 Describes how to view and start/stop system and
network parameters.
“Chapter 5 - Hardware Parameters” on page 49 Describes the hardware installed on your Aastra unit.
“Chapter 6 - Endpoints Configuration” on Describes how to set the administrative state of the
page 53 Aastra unit endpoints.
“Chapter 7 - Syslog Configuration” on page 57 Describes how the Aastra unit handles syslog
messages and notification messages.
“Chapter 8 - Events Configuration” on page 63 Describes how to associate a NOTIFICATION

message and how to send it (via syslog or via a SIP
NOTIFY packet).
Network Parameters
“Chapter 10 - IPv4 vs. IPv6” on page 71 This chapter describes the differences between IPv4
and IPv6 addressing.
“Chapter 11 - Host Parameters” on page 75 Describes how to set the host information used by the
Aastra unit, as well as the default gateway, DNS
servers and SNTP servers configuration source.
“Chapter 12 - Interface Parameters” on page 85 Describes how to set the interfaces of the Aastra unit.
“Chapter 13 - VLAN Parameters” on page 99 Describes how to create and manage dynamic VLANs.
“Chapter 14 - Local QoS (Quality of Service) Describes how to configure packets tagging sent from
Configuration” on page 101 the Aastra unit.
“Chapter 15 - Local Firewall Configuration” on Describes how to configure the local firewall feature.
page 107
“Chapter 16 - IP Routing Configuration” on Describes how to configure the unit's IP routing

page 113 parameters.
“Chapter 17 - Network Firewall Configuration” on Describes how to configure the network firewall
page 121 parameters.
“Chapter 18 - NAT Configuration” on page 127 Describes the configuration parameters to define the
Aastra unit´s NAT.
iv Dgw v2.0 Application

Document Structure Software Configuration Guide
Table 1: Software Configuration Guide Chapter/Appendices (Continued)
Title Summary
“Chapter 19 - DHCP Server Settings” on Describes how to configure the embedded DHCP
page 135 server of the Aastra unit.
POTS Parameters
“Chapter 20 - POTS Configuration” on page 145 Describes how to configure the POTS (Plain Old
Telephony System) line service.
SIP Parameters
“Chapter 21 - SIP Gateways” on page 155 Describes how to add and remove SIP gateways.
“Chapter 22 - SIP Servers” on page 159 Describes how to configure the SIP server and SIP
user agent parameters.
“Chapter 23 - SIP Registration” on page 167 Describes how to configure the registration
parameters of the Aastra unit.
“Chapter 24 - SIP Authentication” on page 179 Describes how to configure authentication parameters
of the Aastra unit
“Chapter 25 - SIP Transport Parameters” on Describes the SIP transport parameters you can set.
page 183
“Chapter 26 - Interop Parameters” on page 189 Describes the SIP interop parameters you can set.
“Chapter 27 - Miscellaneous SIP Parameters” on Describes how to configure the SIP penalty box and
page 211 SIP transport parameters of the Aastra unit.
Media Parameters
“Chapter 28 - Voice & Fax Codecs Configuration” Describes the various voice and fax codecs
on page 231 parameters you can set.
“Chapter 29 - Security” on page 253 Describes how to properly configure the security
parameters of the Aastra unit.
“Chapter 30 - RTP Statistics Configuration” on Describes how to read and configure the RTP
page 257 statistics.
“Chapter 31 - Miscellaneous Media Parameters” Describes how to configure parameters that apply to
on page 263 all codecs.
Telephony Parameters
“Chapter 32 - DTMF Maps Configuration” on Describes how to configure and use the DTMF maps.
page 279
“Chapter 33 - Call Forward Configuration” on Describes how to set and use three types of Call
page 287 Forward.
“Chapter 34 - Telephony Services Configuration” Describes how to set the Aastra unit subscriber
on page 295 services.
“Chapter 35 - Tone Customization Parameters Describes how to override the pattern for a specific
Configuration” on page 317 tone defined for the selected country.
“Chapter 36 - Music on Hold Parameters Describes how to configure the Music on Hold (MOH)
Configuration” on page 321 parameters.
Dgw v2.0 Application v

Preface - About this Manual Document Structure
Table 1: Software Configuration Guide Chapter/Appendices (Continued)
Title Summary
“Chapter 37 - Country Parameters Configuration” Describes how to set the Aastra unit with the proper
on page 325 country settings.
Call Router Parameters
“Chapter 39 - Auto-Routing Configuration” on Describes the call router service.

page 391
“Chapter 39 - Auto-Routing Configuration” on Describes the auto-routing feature.

page 391
Management Parameters
“Chapter 40 - Creating a Configuration Script” Describes how to use the configuration scripts
on page 414 download feature to update the Aastra unit
configuration.
“Chapter 41 - Configuration Backup/Restore” on Describes how to backup and restore the Aastra unit
page 415 configuration.
“Chapter 42 - Firmware Download” on page 423 Describes how to download a firmware pack available
on the designated update files server into the Aastra
unit.
“Chapter 43 - Certificates Management” on Describes how to transfer and manage certificates

page 431 into the Aastra unit.
“Chapter 44 - SNMP Configuration” on page 437 Describes to configure the SNMP privacy parameters
of the Aastra unit.
“Chapter 48 - CWMP Configuration” on page 569 Describes how to set the CWMP parameters of the
Aastra unit.
“Chapter 45 - Access Control Configuration” on Describes how to set the Access Control parameters
page 443 of the Aastra unit.
“File Manager” on page 449 This chapter describes how to use the unit’s File
Manager.
“Chapter 47 - Miscellaneous” on page 451 Describes how to set various parameters used to
manage the Aastra unit.
Appendices
“Appendix A - Country-Specific Parameters” on Lists the various parameters specific to a country such
page 455 as loss plan, tones and rings, etc.
“Appendix B - Scripting Language” on page 477 Describes the Aastra proprietary scripting language. It
also lists a few configuration samples that can be
pasted or typed into the CLI or downloaded into the
Aastra unit via the Configuration Script feature.
“Appendix C - Maximum Transmission Unit Describes the MTU (Maximum Transmission Unit)
(MTU)” on page 485 requirements of the Aastra Unit.
“Appendix D - Web Interface – SNMP Variables Lists the SNMP variables corresponding to the web
Mapping” on page 487 interface of the Aastra unit
vi Dgw v2.0 Application

Document Conventions Software Configuration Guide
Document Conventions
The following information provides an explanation of the symbols that appear on the Aastra unit and in the
documentation for the product.
Warning Definition
Warning: Means danger. You are in a situation that could cause bodily injury. Before you work on any
equipment, you must be aware of the hazards involved with electrical circuitry and be familiar with standard
practices for preventing accidents.
Where to find Translated Warning Definition

For safety and warning information, refer to Appendix A - “Standards Compliance and Safety Information” in
the Aastra unit Hardware Installation Guide. This Appendix describes the international agency compliance and
safety information for the Aastra unit. It also includes a translation of the safety warning listed in the previous
section.
Other Conventions
The following are other conventions you will encounter in this manual.
Caution: Indicates a potentially hazardous situation which, if not avoided, may result in minor or moderate
injury and/or damage to the equipment or property.
Note: Indicates important information about the current topic.
Standards Supported Indicates which RFC, Draft or other standard document is supported for a
specific feature.
SCN vs. PSTN

In Aastra and other vendor’s documentation, the terms SCN and PSTN are used. A SCN (Switched Circuit
Network) is a general term to designate a communication network in which any user may be connected to any
other user through the use of message, circuit, or packet switching and control devices. The Public Switched
Telephone Network (PSTN) or a Private Branch eXchange (PBX) are examples of SCNs.
Standards Supported
When available, this document lists the standards onto which features are based. These standards may be
RFCs (Request for Comments), Internet-Drafts, or other standard documents.
The Dgw v2.0 application’s implementations are based on the standards, so it’s possible that some behaviour
differs from the official standards.
For more information on and a list of RFCs and Internet-Drafts, refer to the IETF web site at http://www.ietf.org.
Dgw v2.0 Application vii

Preface - About this Manual Document Conventions
viii Dgw v2.0 Application

C H A P T E R
1 System Overview
This chapter provides an overview of the Aastra devices supported by the Dgw v2.0 application:
 Introduction to the Aastra devices and the models available.
 Description of the various ways to manage the Aastra unit.
 How to use the DEFAULT/RESET button (partial reset and factory reset procedures).
 How to configure user access to the Aastra unit.
Introduction
The Aastra unit integrates features such as TLS, SRTP, and HTTPS designed to bring enhanced security for
network management, SIP signalling and media transmission aspects.
The following describes the devices that the application supports.
TA7102i
The Aastra TA7102i is a standalone Internet telephony access device that connects to virtually any business
telephone system supporting standard analog lines.
Key Features
The following are the key features offered by the various models available.
Table 2: Aastra Units Key Features
Feature
IP connectivity for analog phones and faxes

Number of simultaneous calls up to 24
FXS interface ports
HTTP, SNMP, FTP and TFTP for configuration and
management
True Plug-and-Play
Automatic configuration script download
Call Routing service
Secure SIP signalling
Secure Media transmission
SNMPv3 and web management
DHCP Client
PPPoE Client
Dgw v2.0 Application 1

Chapter 1 - System Overview DSP Limitation
Table 2: Aastra Units Key Features (Continued)
Feature
T.38 support
Command Line Interface (CLI)
SSL/TLS Encryption
60 VRMS ringing voltage, 2 kilometres loop distance .
DSP Limitation
The Aastra unit models currently suffer from local limitation of their DSPs. When using a codec other than
G.711, enabling Secure RTP (SRTP) and/or using conferences has an impact on the Aastra unit’s overall
performance as SRTP and conferences require CPU power. This means there is a limitation on the lines that
can be used simultaneously, depending on the codecs enabled and SRTP. This could mean that a user picking
up a telephone on these models may not have a dial tone due to lack of resources in order to not affect the
quality of ongoing calls.
The DSPs offer channels as resources to the Aastra unit. The Aastra unit is limited to two conferences per
DSP. See “Conference” on page 305 for more details on Conference limitations.
Howerver, as recomendation is to use the conference service in the call server this would normally not cause
any problem. Please note that:
 One FXS line requires one channel.
 There is a maximum of 2 conferences per DSP
 Each conference requires one additional channel
In the following tables, compressed RTP refers to codecs other than G.711. Numbers in Bold indicate a
possible under-capacity.
TA7102i
Table 3 describes the TA7102 processing capacity.
Table 3: TA7102i Offered Channels vs. Processing Capacity
Offered Chanels Processing Capacity
Model Compr.
Phys. 3-way Conf. G.711 RTP Compr. RTP G.711 SRTP
SRTP
Channels Channels Channels Channels Channels
Channels
TA7102i 2 2 4 4 4 4
2 Dgw v2.0 Application

Management Choices Software Configuration Guide
Management Choices
The Aastra unit offers various management options to configure the unit.
Figure 1: Management Interfaces
Table 4: Management Options
Management Choice Description Features
Web GUI The Aastra unit web interface offers the The Aastra unit web interface allows
following options: you to configure the following
• Password-protected access information:
via basic HTTP • Network attributes
authentication, as described • SIP parameters
in RFC 2617
• VoIP settings
• User-friendly GUI
• Management settings such
as configuration scripts,
restore / backup, etc.
SNMPv1/2/3 The Aastra unit SNMP feature offers the The Aastra unit SNMP feature allows
following options: you to configure all the MIB services.
• Password-protected access
• Remote management
• Simultaneous management
Refer to “Chapter 44 - SNMP
Configuration” on page 437 for more
details.

Chapter 1 - System Overview Management Choices
Table 4: Management Options (Continued)
Management Choice Description Features
Command Line The Aastra unit uses a proprietary CLI The Aastra unit CLI feature allows you
Interface (CLI) to configure all the unit’s parameters. to configure all the MIB services.
Unit Manager The Unit Manager Network (UMN) is a The UMN offers the following:
Network PC-Windows based element • Auto-discovery
management system designed to
• Group provisioning
facilitate the deployment, configuration
and provisioning of Aastra access • SNMP access and remote
devices gateways. management.
The UMN enables the simple and
remote configuration and deployment of
numerous Aastra units.

RESET/DEFAULT Button Software Configuration Guide
RESET/DEFAULT Button
The RESET/DEFAULT button allows you to:

 Cancel an action that was started.
 Revert to known factory settings if the Aastra unit refuses to work properly for any reason or
the connection to the network is lost.
 Reconfigure a unit.
At Run-Time
You can use the RESET/DEFAULT button at run-time – you can press the button while the Aastra unit is
running without powering the unit off. Table 5 describes the actions you can perform in this case.
Table 5: RESET/DEFAULT Button Interaction
RESET/
DEFAULT Button Action Comments LEDs Pattern
Pressed for:
2 to 6 seconds Restarts the Aastra No changes are made to the Power LED:
unit Aastra unit settings. • blinking, 1Hz,
50% duty
All other LEDs:
• OFF
7 to 11 seconds Sets the Aastra unit Sets some of the Aastra unit All LEDs
in Partial Reset Mode configuration to pre-determined • blinking, 1Hz,
values. 50% duty
12 to 16 seconds Restarts the Aastra Deletes the persistent All LEDs
unit in Factory Reset configuration values, creates a • steady ON
new configuration file with the
default factory values, and then
restarts the unit.
17 seconds and No action is taken The RESET/DEFAULT button N/A
more pressed event is ignored.
At Start-Time
You can use the RESET/DEFAULT button at start-time – you power the unit off, and then depress the button
until the LEDs stop blinking and remain ON. This applies the “Factory Reset” procedure (see “Factory Reset”
on page 7). This feature reverts the Aastra unit back to its default factory settings.

Chapter 1 - System Overview RESET/DEFAULT Button
Partial Reset
The Partial reset provides a way to contact the Aastra unit in a known and static state while keeping most of
the configuration unchanged.
Following a partial reset, the Aastra unit management interface is set to the Rescue interface. The default IPv4
address for this interface is 192.168.0.1/24 and has its corresponding link-local IPv6 available and printed on
the sticker under the Aastra unit (see “Chapter 10 - IPv4 vs. IPv6” on page 71 for more details). Any existing
network interface that conflicts with the Rescue interface address is disabled.
You can contact the Aastra unit this address to access its configuration parameters. It is not advised to access
the unit on a regular basis through the Rescue network interface. You should reconfigure the unit’s network
interfaces as soon as possible in order to access it through another interface.
In a partial reset, the following services and parameters are also affected:
 AAA service: User(s) from profile are restored with their factory password.
 SNMP service: Resets the enableSnmpV1, enableSnmpV2, enableSnmpV3 and snmpPort
values to their default values.
 WEB service: Resets the serverPort to its default value.
 CLI service: The CLI variables revert back to their default value.
 NAT service: The configuration is rolled back if it was being modified. A new rule is then
automatically applied in the source and in the destination NAT tables to prevent incorrect rules
from blocking access to the unit. If those rules are not the first priority, they are raised. If there
are no rules in the tables, the new rules are not added since there are no rules to override.
 LFW service: When a partial reset is triggered and the firewall is enabled, the configuration is
rolled back if it was being modified. A new rule is then automatically applied in the firewall to
allow access to the 'Rescue' interface. However, if the firewall is disabled, the configuration is
rolled back but no rule is added.
 HOC service: The Management Interface reverts back to its default value.
 BNI service: The Rescue interface is configured and enabled with:
• its hidden IPv4 link configuration values
• its hidden IPv4 address configuration
• an IPv6 link-local address on all network links
Hidden values are set by the unit's profile.
Just before the Rescue is configured, all IPv4 network interfaces that could possibly conflict with the
Rescue interface are disabled.
If the BNI Service is stopped when the partial reset occurs, it is started and the above configuration
is applied.
 To trigger the Partial Reset:

1. Insert a small, unbent paper clip into the RESET/DEFAULT hole located at the rear of the Aastra
unit. While pressing the RESET/DEFAULT button, restart the unit.
Do not depress before all the LEDs start blinking (between 7-11 seconds).
2. Release the paper clip.
This procedure can also be performed at run-time.
Disabling the Partial Reset

This section describes configuration that is available only in the MIB parameters of the Aastra unit. You can
configure these parameters as follows:
 by using a MIB browser
 by using the CLI
 by creating a configuration script containing the configuration variables

RESET/DEFAULT Button Software Configuration Guide
You can disable the partial reset procedure, even if users depress the Reset/Default button. The following
parameters are supported:
Table 6: Partial Reset Parameters
Parameter Description
All All the actions are allowed: reset, partial reset and factory reset.
DisablePartialReset All actions are allowed except the partial reset.
 The reset action restarts the unit.

 The partial reset action provides a way to contact the unit in a known and static state while
keeping most of the configuration unchanged.
 The factory reset action reverts the unit back to its default factory settings.
 To change the partial reset behaviour:

1. In the hardwareMIB, set the ResetButtonManagement variable to the proper behaviour.
You can also use the following line in the CLI or a configuration script:
hardwareMIB.ResetButtonManagement="Value"
where:
• Value may be as follows:
Table 7: Partial Reset Values
Value Meaning
100 All
200 DisablePartialReset
Factory Reset
The Factory reset reverts the Aastra unit back to its default factory settings. It deletes the persistent MIB values
of the unit, including:
 The firmware pack download configuration files.
 The SNMP configuration, including the SNMPv3 passwords and users.
 The PPPoE configuration, including the PPP user names and passwords.
The Factory reset creates a new configuration file with the default factory values. It should be performed with
the Aastra unit connected to a network with access to a DHCP server. If the unit cannot find a DHCP server,
it sends requests indefinitely.
The following procedure requires that you have physical access to the Aastra unit. However, you can also
trigger a factory reset remotely:
 via the web interface of the Aastra unit. See “Firmware Packs Configuration” on page 425 for
more details.
 via the Command Line Interface of the Aastra unit by using the fpu.defaultsetting
command.
 To trigger the Factory Reset:

1. Power the Aastra unit off.
2. Insert a small, unbent paper clip into the RESET/DEFAULT hole located at the rear of the Aastra
unit. While pressing the RESET/DEFAULT button, restart the unit.
Do not depress before the LEDs stop blinking and are steadily ON. This could take up to 30
seconds.
3. Release the paper clip.
The Aastra unit restarts.

Chapter 1 - System Overview User Access
This procedure resets all variables in the MIB modules to their default value.
When the Aastra unit has finished its provisioning sequence, it is ready to be used with a DHCP-
provided IP address and MIB parameters.
This procedure can also be performed at run-time.
Note: The Factory reset alters any persistent configuration data of the Aastra unit.
User Access
The following describes how to configure user access to the Aastra unit. The access information is available
for the SNMP and Web interface management methods.
Note: Currently, the user name cannot be modified. To access the unit via SNMPv1, you must use the user
name as being the “community name” and there must be no password for this user name.
 To configure the Aastra unit user access:

1. In the aaaMIB, set the password associated with the user name in the usersPassword variable.
aaa.users.Password[UserName="User_Name"]="Value"
Only the “admin” and “public” user names are available for the moment.
2. Set the user name that is used for scheduled tasks in the batchUser variable.
aaa.batchUser="Value"
For instance, if you are using an automatic configuration update everyday at midnight, the relevant
service will use the “batchUser” user to execute the request.
Secure Password Policies

It is possible to validate a password against some password policies to be considered as valid. These policies
may only be activated via customized profiles created by Aastra. The available policies are:
Table 8: Secure Password Policies
Policy Description
Minimum Length of User Password The minimum length the user password must have to be considered
as valid.
Upper and Lower Case Required Indicates if the user password is required to contain an upper and a
on User Password lower case characters to be considered as valid.
Here is an example of a valid password : 'Password' and examples of
invalid passwords : '1234', 'password', '1password', 1PASSWORD.
Numeral character Required on Indicates if the user password is required to contain a numeral
User Password character to be considered as valid.
Here is an example of a valid password : '1password2' and examples
of invalid passwords : 'password', 'Password'.

Where to Go From Here Software Configuration Guide
Table 8: Secure Password Policies (Continued)
Policy Description
Special character Required on Indicates if the user password is required to contain a special character
User Password to be considered as valid.
Here is an example of a valid passwords : 'pass$word', 'pass_word#'
and examples of invalid passwords : 'password', 'Password', '1234',
'1Password'.
For more information on how to get a customized user profile, please refer to your Aastra representative.
Partial Reset
AAA service: User(s) from profile are restored with their factory password.
Where to Go From Here
The current manual offers reference information on the features that the Aastra unit supports.
 If you plan on using the web interface configuration.
 If you plan on using the CLI configuration.
 If you plan on using the SNMP configuration, go to “Chapter 44 - SNMP Configuration” on
page 437
“Appendix B - Scripting Language” on page 477 also offers a few configuration samples that can be pasted or
typed into the CLI or downloaded into the Aastra unit via the Configuration Script feature.

Chapter 1 - System Overview Where to Go From Here

C H A P T E R
2 Command Line Interface (CLI)
This chapter describes how to access the CLI environment in order to perform configuration tasks.
 Introduction
 Configuring the CLI
 Accessing the CLI
• Accessing the CLI via a Telnet Session
• Accessing the CLI via a SSH Session
 Working in the CLI
• Contexts
• Exiting from the CLI
• Command Completion
• Macros
• History
• Service Restart
• Configuring the Aastra unitwith the CLI
 List of Commands / Keywords
Introduction
You can configure the Aastra unit parameters through a proprietary Command Line Interface (CLI)
environment. It allows you to configure the unit parameters by Aastra, Telnet or SSH.
The CLI uses the Aastra proprietary scripting language as described in “Appendix B - Scripting Language” on
page 477.
Configuring the CLI
You must configure the CLI access. This can be done via the MIB variables. Once you have access to the CLI,
you can also use it to configure the access.
 To configure the CLI access:

1. In the cliMIB, set the inactivity expiration delay for exiting the CLI session in the
inactivityTimeOut variable.
If there is no activity during the delay defined, the CLI session is closed. This value is expressed in
minutes.
2. Enable remote Telnet access if applicable by setting the EnableTelnet variable to Enable.
By default, Telnet is not enabled.
3. Set the port on which the Telnet service should listen for incoming Telnet requests in the IpPort.
variable.
4. Enable remote SSH access if applicable by setting the EnableSsh variable to Enable.
5. Set the port on which the SSH service should listen for incoming SSH requests in the IpPort.
variable.

Chapter 2 - Command Line Interface (CLI) Accessing the CLI
The configuration is loaded when it is started. It configures and starts Telnet and SSH according to the options
offered through the configuration variables. The configuration can be updated by the CLI service while running.
Partial Reset
When a partial reset is triggered, the CLI variables revert back to their default value.
Accessing the CLI
You can access the CLI a Telnet or SSH session.

Only one session at a time is allowed. These sections describe how to access the CLI:
 “Accessing the CLI via a Telnet Session” on page 12
• “Opening a Telnet Session with the Unit Manager Network” on page 12
 “Accessing the CLI via a SSH Session” on page 13
Which method you choose depends primarily on your preference and level of experience with one or all of the
options provided. None precludes using other configuration methods. Note that after performing a factory reset
or a firmware update, accessing the CLI may take up to one minute, even if the web and SNMP interfaces are
already accessible.
Note: When performing a partial reset, the root password is removed. See “Partial Reset” on page 6 for
more details.
Accessing the CLI via a Telnet Session

Standards Supported • RFC 854: Telnet Protocol Specification
Connecting via Telnet requires a computer with a Telnet remote client running on a PC that acts as a Telnet
host. The Telnet host accesses the Aastra unit via its LAN or WAN network interface.
 To access the CLI from a remote host using Telnet:

1. Set up the Aastra unit as described in the Hardware Installation Guide.
2. Power on your Aastra unit. Wait 60 seconds before proceeding to the next step.
3. Open a Telnet session to the Aastra unit by using one of the following IP addresses:
• obtained dynamically from the DHCP server
• you have configured statically
• after performing a partial reset (192.168.0.1)
• the link-local IPv6 available and printed on the sticker under the Aastra unit (see
“Chapter 10 - IPv4 vs. IPv6” on page 71 for more details)
If you are using a Telnet port other than 23, (as configured in “Configuring the CLI” on page 11) you
must also specify it.
4. When prompted for a login, type the following:
public
Do not type a password, just press <Enter>. After you successfully connect to the Aastra unit by
using Telnet, you can start using the CLI to configure the unit.
Opening a Telnet Session with the Unit Manager Network

You can use the Aastra Unit Manager Network (UMN) product to launch a Telnet client session to configure
the parameters of the Aastra unit. You can define which Telnet client to use in the UMN.

Accessing the CLI Software Configuration Guide
The Telnet session is opened from the PC where the client application is installed. It thus establishes a direct
connection to the unit. This could cause some problems if the client PC cannot directly access the unit because
of firewall restrictions, etc.
 To open a Telnet session via UMN:

1. In the UMN, autodetect the Aastra unit at one of the IP addresses listed in “Accessing the CLI via a
Telnet Session” on page 12.
Refer to the Unit Manager Network Administration Manual for more details on how to perform this
task.
2. Right-click the unit for which to open a Telnet session.
3. Select the Open Telnet Session option in the context sensitive menu that opens.
The following window opens:
Figure 2: Telnet Session Login
This window may differ if you are not using the default Windows Telnet client.
Accessing the CLI via a SSH Session

Standards Supported • RFC 4251: The Secure Shell (SSH) Protocol Architecture
Connecting via a Secure Socket Shell (SSH) session requires a computer with a SSH or OpenSSH compatible
remote shell client running on a PC that acts as a SSH host. All communication between a client and server
is encrypted before being sent over the network, thus packet sniffers are unable to extract user names,
passwords, and other potentially sensitive data.
 To access the CLI from a remote host using SSH:

1. Set up the Aastra unit as described in the Hardware Installation Guide.
2. Power on your Aastra unit. Wait 60 seconds before proceeding to the next step.
3. Open a SSH session to the Aastra unit by using one of the following IP addresses:
• obtained dynamically from the DHCP server
• you have configured statically
• after performing a partial reset (192.168.0.1)
If you are using a SSH port other than 22, (as configured in “Configuring the CLI” on page 11) you
must also specify it.
4. When prompted for a login, type the following:
public
Do not type a password, just press <Enter>. If you are accessing the unit through the CLI for the
first time or after a factory reset, you may be presented with a warning message regarding the unit’s
identification. You can accept the message and continue.
After you successfully connect to the Aastra unit by using Telnet, you can start using the CLI to
configure the Aastra unit.

Chapter 2 - Command Line Interface (CLI) Working in the CLI
Working in the CLI
The command interpreter interface of the CLI is a program called by the Telnet or SSH client.
It allows you to browse the parameters of the unit. It also allows you to write the command lines and the CLI
interprets and executes it. The following figure illustrates the CLI in the global context after preforming a ls
command:
Figure 3: CLI Global Context
Contexts
The CLI has various contexts. A context is defined as a service name indicated by its textual key (for instance,
the Conf service). Upon entering the CLI, you are located in the Global context. This is indicated by the
following prompt:
Global>
You can change context by using the cd (change directory) command with the following syntax:
cd Service_Name
This allows you to enter into a service context. You can thus execute commands without writing the service
name. For instance:
Global>cd Conf
The prompt then changes to:
Conf>
You can use the following to get back to the global context:
Conf>cd
You can also access another service context from the Conf context:
Conf>cd Bni
Executing a command is different depending on if you are in the global context or a service context. See
“commands” on page 20 for examples.
Exiting from the CLI

To exit the CLI, type the exit command from the global context or any of the service contexts.

Working in the CLI Software Configuration Guide
Command Completion
The CLI command completion function works on everything in the CLI including aliases, macros, commands,
names, etc. It is case insensitive, which means that typing interface is the same as typing Interface.
However, names that are unique are case sensitive, such as interface names.
To display all possible commands or statements, enter at least one character and press the Tab key to
complete the command line. If more than one possibility exists, they are listed and you can select the one you
want.
Let’s say for instance that you type the following command in the Bni context:
Bni>Net[+ Tab key]
The CLI displays the following choices:
NetworkInterfaces NetworkInterfacesStatus
Bni>NetworkInterfaces
Macros
Macros are internal hardcoded commands that are frequently used. The CLI currently supports the following
macros:
Table 9: Macros
Macro Description
Reboot Reboots the unit.

Restart Restarts the services when in a service context or restarts the unit in the global context.
For instance:
Global>Mipt.Restart
You can see the list of available macros by typing the following command from anywhere in the CLI:
Conf.Macros
This will return a table similar to the following:
_____________________________
| Name | Description |
|__________|__________________|
| Reboot | Reboot unit |
| Restart | Restart service |
|__________|__________________|
History
You can recall the history commands and navigate through the history using the up and down arrows.
Services Restart
Whenever you perform changes in the configuration, this usually means that you must restart a service for the
changes to take effect. When this is the case, the following message appears in the CLI:
Need Restart
Use the Restart macro as described in “Macros” on page 15.
Syslog Messages
You can access the notifications, diagnostic traces and SIP signalling logs of the Aastra unit. Use the logs on
command to display Syslog traces as soon as they are sent. Use the logs off command to stop displaying
the logs.

Chapter 2 - Command Line Interface (CLI) Working in the CLI
Configuring the Aastra unit with the CLI

Once you are in the CLI, you can configure all the parameters of the Aastra unit with the various keywords
available. These keywords are described in “List of Commands / Keywords” on page 18. You must however
have a good understanding of the parameters structure
A good way of working with the CLI is to create the complete configuration in a text file, then copy and paste
chunks of the configuration in the CLI. This avoids to type all the commands in the CLI itself. However, be
aware that you must not copy configuration when a service needs to be restarted. You must first restart the
service before continuing.
Let’s say for instance you are in the Global context and you want to see the inactivity timeout value of the CLI.
Type the following:
Global>get Cli.InactivityTimeOut
The CLI displays the current value. If you want to change this value to 10 minutes, type the following:
Global>set Cli.InactivityTimeOut=10
Refer to “Appendix B - Scripting Language” on page 477 for samples of configurations you can use in the CLI.
The samples include the configuration required to perform a basic call between an ISDN telephone and an
analog telephone. These samples may also be used in configuration scripts that you can download into the
Aastra unit.
Current Unit Status

The current unit status is displayed every time a user is authenticated by the CLI. You can also display the
same information during a session by executing the sysinfo command. The information displayed is:
 System Description
 Serial Number
 Firmware Version
 Host Name
 Mac Address
 System Uptime
 System Time
 Snmp Port
 Installed Hardware Information (Name, description, location).
Welcome Message
You can define a message that is displayed when connecting to the CLI by typing the following:
Global>set Cli.WelcomeMessage=Value
Where Value is the actual message you want displayed. The following escape characters are supported:
 \n for new line
 \t for tab
 \\ for the \ character.
Other characters are left unchanged.
Help
The CLI allows you to get help on the various keywords supported. You can have access to general or
contextual help.
You can access the general help by typing the help keyword:
Global>help
In that case, the CLI displays the list of all keywords available.

Working in the CLI Software Configuration Guide
Figure 4: CLI Global Help
You can also access a more specific general help by typing the help keyword in a context.
Conf>help
In that case, the CLI displays the list of all keywords available as well as a description of the context.
Figure 5: CLI Global Help Variation
You can access the contextual help by typing the help keyword followed by the keyword.
Global>help set

Chapter 2 - Command Line Interface (CLI) List of Commands / Keywords
Figure 6: CLI Contextual Help
See “List of Commands / Keywords” on page 18 for a list of keywords available.

Finally, you can access a more specific contextual help by typing the help keyword followed by the name of
the expression (scalar, table, command, column, service).
Global>help Cli.InactivityTimeOut
Figure 7: CLI Expression Help
List of Commands / Keywords
The following sections describe the commands and keywords and their syntax depending on the context in
which you are located. Each syntax also has an example in blue.
access
Retrieves the access type of the expression. The expression
may be a variable (scalar), a table cell, or a table column. Applies To
Services Tables Columns Variables

Variables – Global Context
Use this syntax when in the global context.
access Service_Name.Scalar_Name
access Cli.InactivityTimeOut
Variables – Service Contexts

Use this syntax when in a service context.
access Scalar_Name
access InactivityTimeOut
Table Cell or Column Properties – Global Context

Use this syntax when in the global context. The Index parameter is the first column of the table.
access Service_Name.Table_Name[Index=key].Column_Name
access Hoc.DnsServersInfo[Priority=2].IpAddress
access Service_Name.Table_Name.Column_Name

List of Commands / Keywords Software Configuration Guide
access Hoc.DnsServersInfo.IpAddress
Table Cell or Column Properties – Service Contexts

Use this syntax when in a service context. The Index parameter is the first column of the table.
access Table_Name[Index=key].Column_Name
access DnsServersInfo[Priority=2].IpAddress
access Table_Name.Column_Name
access DnsServersInfo.IpAddress
alias / unalias
The alias function allows you to create a keyboard shortcut, an
abbreviation, a mean of avoiding typing a long command Applies To
sequence. You can assign an alias to services, scalars, tables,
and commands. You cannot currently assign an alias to Services Tables Columns Variables
columns.
Once an alias has been added, you can use it in place of the
entity name when typing commands. You can delete an alias
with the unalias command.
Note: When naming an alias, you cannot use an existing macro name, service name, nor MIB object name
from the same context.
You can see the list of available aliases by typing the following command from anywhere in the CLI:
Conf.Alias
This will return a table similar to the following:
_______________________________________________
| Name | Entity | Type | Context |
|__________|___________________|______|_________|
| TimeOut | InactivityTimeOut | 200 | Cli |
|__________|___________________|______|_________|

alias Service_Name.Scalar_Name=aliasName
alias Cli.InactivityTimeOut=timeout
unalias aliasName
unalias timeout

alias Scalar_Name = aliasName
alias InactivityTimeOut=timeout
unalias aliasName
unalias timeout
Tables or Columns – Global Context

alias Service_Name.Table_Name=aliasName
alias Hoc.DnsServersInfo.IpAddress
alias Service_Name.Table_Name.Column_Name=aliasName
alias Hoc.DnsServersInfo.IpAddress = IPAddress
unalias aliasName
unalias IPAddress

Tables or Columns – Service Contexts

alias Table_Name = aliasName
alias DnsServersInfo.IpAddress
alias Table_Name.Column_Name=aliasName
alias DnsServersInfo.IpAddress=IPAddress
unalias aliasName
unalias IPAddress
cd
Changes context (global or service context).
Applies To
Enter into a Context – Global Context
cd Service_Name
cd Hoc
Enter into a Context – Service Contexts

cd Service_Name
cd Hoc
Get Back to the Global Context from a Service Context

cd
columnars
Retrieves the columns associated with a table.
Applies To
Table Consultation – Global Context
columnars Service_Name.Table_Name
columnars Hoc.DnsServersInfo
Table Consultation – Service Contexts

columnars Table_Name
columnars DnsServersInfo
commands
Retrieves the commands associated with a service or a table.
Applies To
Service Consultation – Global Context
commands Service_Name
commands Bni
Service Consultation – Service Contexts


commands

commands Service_Name.Table_Name
commands Hoc.DnsServersInfo

commands Table_Name
commands DnsServersInfo
defval
Retrieves the default value of the expression. The expression
may be a variable (scalar), a table column, or a table cell). Applies To

defval Service_Name.Scalar_Name
defval Cli.InactivityTimeOut

defval Scalar_Name
defval InactivityTimeOut
Cell or Column Properties – Global Context

defval Service_Name.Table_Name[Index=key].Column_Name
defval Hoc.DnsServersInfo[Priority=2].IpAddress
defval Service_Name.Table_Name.Column_Name
defval Hoc.DnsServersInfo.IpAddress
Cell or Column Properties – Service Contexts

defval Table_Name[Index=key].Column_Name
defval DnsServersInfo[Priority=2].IpAddress
defval Table_Name.Column_Name
defval DnsServersInfo.IpAddress
dump
Displays the unit’s whole configuration on screen.
dump Applies To

get
Retrieves the value of the expression. The expression may be a
variable (scalar), a table, a table row, a table column, or a table Applies To
cell). Note that entering the get command is optional.
Variable Consultation – Global Context

Service_Name.Scalar_Name
Cli.InactivityTimeOut
get Service_Name.Scalar_Name
get Cli.InactivityTimeOut
Variable Consultation – Service Contexts

Scalar_Name
InactivityTimeout
get Scalar_Name
get InactivityTimeOut

Service_Name.Table_Name
Hoc.DnsServersInfo
get Service_Name.Table_Name
get Hoc.DnsServersInfo

Table_Name
DnsServersInfo
get Table_Name
get DnsServersInfo
Column Consultation – Global Context

Service_Name.Table_Name.Column_Name
Hoc.DnsServersInfo.Priority
get Service_Name.Table_Name.Column_Name
get Hoc.DnsServersInfo.Priority
Column Consultation – Service Contexts

Table_Name.Column_Name
DnsServersInfo.Priority
get Table_Name.Column_Name
get DnsServersInfo.Priority
Row Consultation – Global Context

Service_Name.Table_Name[Index=key]
Hoc.DnsServersInfo[Priority=2]
get Service_Name.Table_Name[Index=key]
get Hoc.DnsServersInfo[Priority=2]

Row Consultation – Service Contexts

Table_Name[Index=key]
DnsServersInfo[Priority=2]
get Table_Name[Index=key]
get DnsServersInfo[Priority=2]
Cell Consultation – Global Context

Service_Name.Table_Name[Index=key].Column_Name
Hoc.DnsServersInfo[Priority=2].IpAddress
get Service_Name.Table_Name[Index=key].Column_Name
get Hoc.DnsServersInfo[Priority=2].IpAddress
Cell Consultation – Service Contexts

Table_Name[Index=key].Column_Name
DnsServersInfo[Priority=2].IpAddress
get Table_Name[Index=key].Column_Name
get DnsServersInfo[Priority=2].IpAddress
help
Retrieves the documentation related to the expression. This
keyword is case sentitive. Applies To
You can have access to general or contextual help. You can
access the general help by typing the help keyword. You can
access the contextual help by typing the help keyword followed
by the name of the expression.

help Service_Name
help Bni

help

help Service_Name.Table_Name
help Hoc.DnsServersInfo
help Service_Name.Table_Name.Column_Name
help Hoc.DnsServersInfo.IpAddress

help Table_Name
help DnsServersInfo
help Table_Name.Column_Name

help DnsServersInfo.IpAddress
Commands – Global Context

help Service_Name.Command
help Scm.LockConfig
Commands – Service Contexts

help Command
help LockConfig
indexes
Retrieves the indexes associated with the expression of a table.
The expression may be the table itself or one of its columns. Applies To

indexes Service_Name.Table_Name
indexes Hoc.DnsServersInfo

indexes Table_Name
indexes DnsServersInfo

indexes Service_Name.Table_Name.Column_Name
indexes Hoc.DnsServersInfo.IpAddress

indexes Table_Name.Column_Name
indexes DnsServersInfo.IpAddress
keys
Retrieves the keys associated with the expression of a table. The
expression may be the table itself or one of its columns. Applies To

keys Service_Name.Table_Name
keys Hoc.DnsServersInfo

keys Table_Name
keys DnsServersInfo


keys Service_Name.Table_Name.Column_Name
keys Hoc.DnsServersInfo.IpAddress

keys Table_Name.Column_Name
keys DnsServersInfo.IpAddress
logs off
Stops to display of Syslog traces.
logs off Applies To
logs on
Displays Syslog traces as soon as they are sent.
The traces displayed are the notifications coming from the Applies To
services, the diagnostic traces and the Signaling Logs.
logs on
ls
Retrieves the list of services available in a global context and the
objects of the service on a service context. Applies To

ls Service_Name
ls Bni

ls

ls Service_Name.Table_Name
ls Hoc.DnsServersInfo

ls Table_Name
ls DnsServersInfo

name
Retrieves the name of the expression. This keyword is case
sensitive. You must type the exact name after the name key. Applies To

Variables Configuration – Global Context
name Service_Name.Scalar_Name
name Cli.InactivityTimeOut
Variables Configuration – Service Contexts

name Scalar_Name
name InactivityTimeOut
Table Configuration – Global Context

name Service_Name.Table_Name
name Hoc.DnsServersInfo
name Service_Name.Table_Name.Column_Name
name Hoc.DnsServersInfo.IpAddress
Table Configuration – Service Contexts

name Table_Name
name DnsServersInfo
name Table_Name.Column_Name
name DnsServersInfo.IpAddress
Command Execution – Global Context

name Service_Name.Command
name Scm.LockConfig
Command Execution – Service Contexts

name Command
name LockConfig
objects
Retrieves the objects associated with the expression of a
service. Applies To

objects Service_Name
objects Bni


objects
Table consultation – Global Context

objects Service_Name.Table_Name
objects Hoc.DnsServersInfo
Table consultation – Service Contexts

objects Table_Name
objects DnsServersInfo
PCapture
Starts a network capture. Typing ctrl+c stops immediately a running capture command and displays statistics.
Supported parameters can be found by typing "help pcapture".
The Telnet and SSH ports are automatically filtered out. The host addresses are not converted to names to
avoid DNS lookups. The protocol and port numbers are not converted to names either.
Use this syntax.
pcapture [options] [expression]
pcapture -raw -c 50 port 161
Options:
 -c 'count': Exit after receiving 'count' packets.
 -raw: Raw packets are output (unreadable output, must be redirected to file or Wireshark)
 -D: Print the list of network interfaces available on the system and on which pcapture can
capture packets.
 -e: Print the link-level header on each dump line.
 -i 'if': Listen on interface 'if'. Can be any of the interfaces returned by option -D or can be set to
'any'. 'any' will listen on all interfaces but not in promiscuous mode.
 -p: Don't put the interface into promiscuous mode.
 -S: Print absolute, rather than relative, TCP sequence numbers.
 -T 'expression': Force packets selected by 'expression' to be interpreted of the specified type.
Supported types are rtp, rtcp, snmp, tftp.
Expression:
 Selects which packets will be dumped. If no expression is given, all packets on the net will
be dumped. Otherwise, only packets for which expression is 'true' will be dumped. For the
expression syntax, see pcap-filter(7).
It is possible to route the capture to Wireshark to have a remote live capture. From the remote PC (Windows
or Linux), type the following command:
plink.exe -pw "" [email protected] "pcapture -raw port 161" | wireshark -k -i -
This example connects by using plink (from putty) in SSH to the unit 10.4.127.128 by using the username
"public" and an empty password. It would capture the SNMP packets.
For more information in the pcapture command, please refer to the following page: http://www.tcpdump.org/
pcap3_man.html.
ping (IPv4)
Executes a ping command using IPv4 with the arguments and the target host provided by the user.
Use this syntax when using the ping command:
ping [-c COUNT -s SIZE -q] host_name
ping -c 3 -s 300 -q 192.168.0.25

The supported ping arguments are:

 -c COUNT: Stops the ping after it has sent COUNT packets.
 -s SIZE: Sends SIZE data bytes in packets (default = 56).
 -q: Shows information only at the start and when finished.
Typing Ctrl+c immediately stops a running ping command and displays statistics.
ping (IPv6)
Executes a ping command using IPv6 with the arguments and the target host provided by the user.
Use this syntax when using the ping command:
ping [-c COUNT -s SIZE -q] host_name
ping -c 3 -s 300 -q 192.168.0.25
The supported ping arguments are:
 -c COUNT: Stops the ping after it has sent COUNT packets.
 -s SIZE: Sends SIZE data bytes in packets (default = 56).
 -q: Shows information only at the start and when finished.
Typing Ctrl+c immediately stops a running ping command and displays statistics.
scalars
Retrieves the scalars associated with the expression of a
service. Applies To

scalars Service_Name
scalars Bni

scalars
set
Assigns a constant value to the expression. The expression may
be a variable (scalar) or a table cell. Applies To

Service_Name.Scalar_Name = constant
Cli.InactivityTimeOut = 25
set Service_Name.Scalar_Name = constant
set Cli.InactivityTimeOut = 25

Scalar_Name = constant
InactivityTimeOut = 25
set Scalar_Name = constant
set InactivityTimeOut = 25
Cell Configuration – Global Context

Service_Name.Table_Name[Index=key].Column_Name=Value
Hoc.DnsServersInfo[Priority=2].IpAddress=”192.168.0.10”
set Service_Name.Table_Name[Index=key].Column_Name=Value
set Hoc.DnsServersInfo[Priority=2].IpAddress=”192.168.0.10”
Cell configuration – Service Contexts

Table_Name[Index=key].Column_Name=Value
DnsServersInfo[Priority=2].IpAddress=”192.168.0.10”
set Table_Name[Index=key].Column_Name=Value
set DnsServersInfo[Priority=2].IpAddress=”192.168.0.10”
show
Retrieves the value of the expression.
Applies To
Variable Consultation – Global Context
show Service_Name.Scalar_Name
show Cli.InactivityTimeOut
Variable Consultation – Service Contexts

show Scalar_Name
show InactivityTimeOut

show Service_Name.Table_Name
show Hoc.DnsServersInfo

show Table_Name
show DnsServersInfo

show Service_Name.Table_Name.Column_Name
show Hoc.DnsServersInfo.IpAddress

show Table_Name.Column_Name
show DnsServersInfo.IpAddress
Row Consultation – Global Context

show Service_Name.Table_Name[Index=key]
show Hoc.DnsServersInfo[Priority=2]

Row Consultation – Service Contexts

show Table_Name[Index=key]
show DnsServersInfo[Priority=2]
Cell Consultation – Global Context

show Service_Name.Table_Name[Index=key].Column_Name
show Hoc.DnsServersInfo[Priority=2].IpAddress
Cell Consultation – Service Contexts

show Table_Name[Index=key].Column_Name
show DnsServersInfo[Priority=2].IpAddress
sysinfo
Displays the current unit status. The information displayed is:
 System Description
 Serial Number
 Firmware Version
 Host Name
 Mac Address
 System Uptime
 System Time
 Snmp Port
 Installed Hardware Information (Name, description, location).
tables
Retrieves the tables associated with the a service.
Applies To
tables Service_Name
tables Bni

tables
type
Retrieves the type of the data of the expression. The expression
may be a variable (scalar), a table column, or a table cell. Applies To

type Service_Name.Scalar_Name
type Cli.InactivityTimeOut


type Scalar_Name
type InactivityTimeOut
Cell or Column Properties – Global Context

type Service_Name.Table_Name[Index=key].Column_Name
type Hoc.DnsServersInfo[Priority=2].IpAddress
type Service_Name.Table_Name.Column_Name
type Hoc.DnsServersInfo.IpAddress
Cell or Column Properties – Service Contexts

type Table_Name[Index=key].Column_Name
type DnsServersInfo[Priority=2].IpAddress
type Table_Name.Column_Name
type DnsServersInfo.IpAddress
Command Execution
This section describes the syntax to use to execute a MIB command.
Global Context
Service_Name.Command arg1=value1 -b arg2=[value2 value3 value4]
SipEp.InsertGateway Name=test
Service Context
Command arg1=value1 -b arg2=[value2 value3 value4]
InsertGateway Name=test


C H A P T E R
3 Web Interface Configuration
The Aastra unit contains an embedded web server to set parameters by using the HTTP or HTTPS protocol.
Standards Supported • RFC 1945: Hypertext Transfer Protocol - HTTP/1.0

• RFC 2616: Hypertext Transfer protocol - HTTP/1.1.
This chapter describes the following:

 Introduction to the Aastra unit web pages.
 Short description of the Aastra unit SNMP configuration.
 How to access the web interface and description of the various menus available.
 How to submit changes.
Introduction
The web interface may be used to:

 View the status of the Aastra unit.
 Set the uplink parameters of the Aastra unit.
 Perform a firmware update, configuration scripts download, or configuration backup/restore.
 Set numerous parameters of the Aastra unit.
All of the parameters in the web interface may also be configured via SNMP. See“Chapter 44 - SNMP
Configuration” on page 437 for more details.
 To configure the web-based configuration service:

1. In the webMIB, locate the serverGroup folder.
2. Define the HTTP mode(s) to which the Web server should listen in the httpMode variable.
web.httpMode="Value"
where Value may be as follows:
Table 10: HTTP Modes
Value Mode Description
100 Secure The Web server only accepts requests using HTTPS. Requests using HTTP
are ignored. This is the default value.
200 Unsecure The Web server only accepts requests using HTTP. Requests using HTTPS
are ignored.
300 Both The Web server accepts requests using HTTP or HTTPS.
If you are using HTTPS (either in “Secure” mode or “Both” mode), the web server needs a valid
server certificate with “server authentication” extended key usage installed on the Aastra unit. See
“Chapter 43 - Certificates Management” on page 431 for more details.
Accessing the web pages via HTTPS adds additional delay since encryption is used. To access the
unit via HTTPS, your browser must support RFC 2246 (TLS 1.0).

Chapter 3 - Web Interface Configuration Introduction
Note that the web server does not listen to the configured modes when the management interface
is down or a configuration error occurred (e.g., missing or invalid certificate for HTTPS mode) while
setting up the web server.
3. Set the TCP port on which the web service listens for HTTP requests in the serverPort variable.
web.serverPort="Value"
4. Set the port on which the web service listens for HTTPS requests in the secureServerPort
variable.
web.secureServerPort="Value"
HTTP User-Agent Header Format

You can define the text to display in the HTTP User-Agent header. You can use macros to include information
specific to the unit.
You can also define the same information in the SIP User-Agent header. See“SIP User-Agent Header Format”
on page 208 for more details.
 To set the HTTP User-Agent header format:

1. In the hocMIB, set the HTTP User-Agent header format in the httpUaHeaderFormat variable.
hoc.httpUaHeaderFormat="Value"
where Value may contain any text, as well as one or more of the following macros:
Table 11: Macros Supported
Macro Description
%version% Application version.

%mac% MAC address.
%product% Product name.
%profile% Profile.
%% Insert the % character.
For instance, the default value is:

%product%/v%version% %profile%

Using the Web Interface Software Configuration Guide
Using the Web Interface
Aastra recommends that you use the latest version of the Microsoft
® Internet Explorer web browser to
properly access the web interface.
 To use the web interface configuration:

1. In your web browser’s address field, type the IP address of the Aastra unit LAN interface (if you have
performed a partial reset, this is 192.168.0.10).
Figure 8: Login Window
2. Enter the proper user name and password.

The user name and password are case sensitive hence they must be entered properly. Default
factory values are:
• User Name: admin
• Password: administrator
You can also enter the user name public and no password.
3. Click Login.
The Information web page displays. It stays accessible for as long as the Internet browser used to
access the Aastra unit web interface is opened.
Figure 9: Information Web Page
4. Click Sign Out to end your Aastra web session.

The Login Window web page displays.

Chapter 3 - Web Interface Configuration Using the Web Interface
Menu Items
The Menu frame is displayed at the top of the browser window. It contains management links that allow you
to display web pages in the Content frame. The management links available vary depending on the Aastra unit
you are using.
Table 12: Menu Frame Links
Link Description
Information: Displays, in read-only format, the status of the Aastra unit.

Services: Allows you to start/stop the services running on the Aastra unit. See
“Chapter 4 - Services” on page 43 for more details.
Hardware: Not applicable for TA7102i.
Endpoints: Allows you to configure the administrative state of the Aastra unit’s
ports. See“Chapter 6 - Endpoints Configuration” on page 53 for more details.
System Syslog: Allows you to configure the Aastra unit to properly handle syslog
messages and notification messages. See “Chapter 7 - Syslog Daemon
Events: Allows you to associate a NOTIFICATION message and how to send it
(via syslog or via a SIP NOTIFY packet). See“Chapter 8 - Notification Events” on
page 63 for more details.
Local Log: Displays local log status and entries of your Aastra unit. See “Chapter
9 - Local Log” on page 81 for more details.
Status: Displays, in read-only format, the network parameters status of the Aastra
unit.
Host: Allows you to configure the host parameters of the Aastra unit. See
“Chapter 11 - General Configuration” on page 75 for more details.
Interfaces: Allows you to configure the uplink information used by the Aastra unit.
See “Chapter 11 - General Configuration” on page 75 for more details.
VLAN: Allows you to create and manage VLANs on the Aastra unit. See “Chapter
13 - VLAN Parameters” on page 99 for more details.
QoS: Allows you to configure packets tagging sent from the Aastra unit. See
“Chapter 14 - Local QoS (Quality of Service) Configuration” on page 101 for more
details.
Network Local Firewall: Allows you to configure the local firewall service of the Aastra
unit. See“Chapter 15 - Local Firewall Configuration” on page 107 for more details.
IP Routing: Allows you to configure the IP routing parameters of the Aastra unit.
See “Chapter 16 - IP Routing Configuration” on page 127 for more details.
Network Firewall: Allows you to configure the network firewall service of the
Aastra unit. See “Chapter 17 - Managing the Network Firewall” on page 121 for
more details.
NAT: Allows you to configure the NAT service of the Aastra unit. See “Chapter 18
- NAT Configuration” on page 127 for more details.
DHCP Server: Allows you to configure the the embedded DHCP server of the
Aastra unit. See “Chapter 19 - DHCP Server Settings” on page 135 for more
details.

Using the Web Interface Software Configuration Guide
Table 12: Menu Frame Links (Continued)
Link Description
Status: Allows you to view the status of the Aastra unit POTS parameters. See
“Chapter - POTS Parameters” on page 143for more details.
Config: Allows you to configure the POTS parameters of the Aastra unit. See
POTS “Chapter 20 - General POTS Configuration” on page 145 for more details.
FXS Config: Allows you to configure the FXS parameters of the Aastra unit. See
“Chapter 20 - POTS Configuration” on page 145 for more details.
FXO Config: Not applicable.
Gateways: Allows you to add and remove SIP gateways in the Aastra unit. See
“Chapter 21 - SIP Gateways Configuration” on page 155 for more details.
Servers: Allows you to configure the SIP server and SIP user agent parameters
of the Aastra unit. See “Chapter 22 - SIP Servers” on page 159 for more details.
Registrations: Allows you to configure the registration parameters of the Aastra
unit. See “Chapter 23 - Endpoints Registration” on page 167 for more details.
Authentication: Allows you to configure authentication parameters of the Aastra
unit. See “Chapter 24 - SIP Authentication” on page 179 for more details.
SIP Transport: Allows you to configure the SIP transport parameters of the Aastra
unit. See “Chapter 25 - SIP Transport Parameters” on page 183 for more details.
Interop: Allows you to configure the SIP interop parameters of the Aastra unit.
See “Chapter 26 - Interop Parameters” on page 189 for more details.
Misc: Allows you to configure interoperability features of the Aastra unit. See
“Chapter 27 - SIP Penalty Box” on page 211 for more details.
Codecs: Allows you to configure the voice and data codec related parameters of
the Aastra unit. See “Chapter 28 - Voice & Fax Codecs Configuration” on
Security: Allows you to properly configure the security parameters of the Aastra
unit. See “Chapter 29 - Security” on page 253 for more details.
Media
RTP Stats: Allows you to read and configure the RTP statistics collected by the
Aastra unit. See “Chapter 30 - RTP Statistics Configuration” on page 257 for
more details.
Misc: Allows you to configure parameters that apply to all codecs. See “Chapter
31 - Miscellaneous Media Parameters” on page 263 for more details.
DTMF Maps: Allows you to configure the various DTMF maps of the Aastra unit.
See “Chapter 32 - DTMF Maps Configuration” on page 279 for more details.
Call Forward: Allows you to configure three types of Call Forward. See “Chapter
33 - Call Forward Configuration” on page 287 for more details.
Services: Allows you to configure the Aastra unit subscriber services. See
“Chapter 34 - General Configuration” on page 295 for more details.
Tone Customization: Allows you to override the pattern for a specific tone
defined for the selected country. See “Chapter 35 - Tone Customization
Parameters Configuration” on page 317 for more details.
Telephony Music on Hold: Allows you to configure the Music on Hold service of the Aastra
unit. See “Chapter 36 - Configuring the TFTP Server” on page 321 for more
details.
Misc: Allows you to configure the country in which the Aastra unit is located. See
“Chapter 37 - Country Configuration” on page 325 for more details.

Chapter 3 - Web Interface Configuration Submitting Changes
Table 12: Menu Frame Links (Continued)
Link Description
Status: Allows you to view the current status of the call routing service. See
“Chapter 38 - Call Router Configuration” on page 335 for more details.
Route Config: Allows you to configure the call routing service of the Aastra unit.
Call Router
See “Chapter 38 - Call Router Configuration” on page 335 for more details.
Auto-routing: Allows you to configure the auto-routing feature of the Aastra unit.
See “Chapter 39 - Auto-Routing Configuration” on page 391 for more details.
Configuration Scripts: Allows you to configure the various configuration scripts
parameters of the Aastra unit. See “Chapter 40 - Creating a Configuration Script”
Backup / Restore: Allows you to configure how to backup and restore the Aastra
unit’s configuration. See “Chapter 41 - Configuration Backup/Restore” on
Firmware Upgrade: Allows you to configure the various firmware upgrade
parameters of the Aastra unit. See “Chapter 42 - Firmware Download” on
Certificates: Allows you to add and delete security certificates in the Aastra unit.
Management See “Chapter 43 - Certificates Management” on page 431 for more details.
SNMP: Allows you to configure the SNMP privacy parameters of the Aastra unit.
See “Chapter 44 - SNMP Configuration” on page 437 for more details.
CWMP: Not applicable.
Access Control: Allows you to set the Access Control parameters of the Aastra
unit. See “Chapter 45 - Users” on page 443 for more details.
File:Allows you to use the unit’s File Manager. See “File Manager” on page 449
for more details.
Misc: Allows you to set various parameters used to manage the Aastra unit. See
“Chapter 47 - Management Interface Configuration” on page 451 for more details.
Reboot Allows you to restart the Aastra unit.
Submitting Changes
When you perform changes in the web interface and click the Submit button, the Aastra unit validates the
changes. A message is displayed next to any invalid value. A message is also displayed if a service must be
restarted and a link is displayed at the top of the page. This link brings you to the Services page. In this page,
each service that requires to be restarted has a “*” beside its name. See “Chapter 4 - Services” on page 53
for more details.
If you are not able to restart one or more services, click the Reboot link in the top menu. The Reboot page then
opens. You must click Reboot. This restarts the Aastra unit. If the unit is in use when you click Reboot, all calls
are terminated.

Where to Go From Here? Software Configuration Guide
Where to Go From Here?
If you want to configure the Aastra unit to perform a basic call, this usually involves the following:
Table 13: Basic Call Configuration Steps
Action Description Where to?
Configuring the POTS You must minimally configure the FXS “Chapter 20 - POTS
parameters interfaces so that they can send and receive Configuration” on page 145
TA7102i calls.
Configuring the SIP Configuring the SIP endpoint allows you to “Chapter 22 - Introduction” on
Endpoint register your ISDN telephone or FXS interfaces page 159
to a SIP server. This includes setting the “Chapter 23 - Registration
following parameters: Configuration” on page 169
• Registrar Server Host “Chapter 21 - SIP Gateways” on
• Proxy Home Domain Host page 155
• User Name
• Friendly Name
• Gateway Name
Configuring the Call You must create routes that will route calls from “Call Router Configuration” on
Router with Routes FXS to SIP and from SIP to FXS. page 335
Configuration of the You must create mappings that will allow you to “Mappings” on page 358
Call Router: Mapping properly communicate from FXS to SIP and
from SIP to FXS.
Using Secure Communication

The Aastra unit allows you to use a secure communication whenever required. You must set the Aastra unit
with security parameters:
Table 14: Secure Communication Steps
Step Where to?
5. Transfer a valid CA certificate into the Aastra “Chapter 46 - Certificates Management” on

unit. page 557
6. Use secure signalling by enabling the TLS “Chapter 25 - SIP Transport Parameters” on
transport protocol. page 183
7. Use secure media by:
• Defining the SRTP/ SRTCP “Base Ports Configuration” on page 273
base port.
• Setting the RTP secure mode
“Security Parameters” on page 253
to “Secure” or “Secure with
fallback”.

Chapter 3 - Web Interface Configuration Where to Go From Here?

System Parameters
Page Left Intentionally Blank
C H A P T E R
4 Services
This chapter describes how to view and start/stop system and network parameters of the Aastra unit.
Services Table
The Aastra unit uses many services grouped in two classes: system and user. You can perform service
commands on user services, but not the system services.
Whenever you perform changes in the various sections of the web interfaces, this usually means that you must
restart a service for the changes to take effect. When a service needs to be restarted, it is displayed in bold
and the message Restart needed is displayed in the Comment column.
If you are not able to restart a service because it is a system service, click the Reboot link in the top menu.
The Reboot page then opens. You must click Reboot. This restarts the Aastra unit. If the unit is in use when
you click Reboot, all calls are terminated.
 To manage the Aastra unit services:

1. In the web interface, click the System link, then the Services sub-link.
Figure 10: System – Services Web Page

Chapter 4 - Services Services Table
The following are the services available.

Table 15: Aastra unit Services
Service Description
System Services
Authentication, Authorization Authenticates a user and grants rights to perform specific tasks
and Accounting (AAA) on the system.
Certificate Manager (CERT) Manages certificate files and provides access to these
certificates.
Configuration Manager Responsible of configuration scripts transfers, as well as
(CONF) configuration image upload/download for backup/restore of the
unit configuration.
Device Control Manager Auto-detects and identifies the hardware components of the unit.
(DCM)
Ethernet Manager (ETH) Configures the system's Ethernet ports parameters.
File Manager (FILE) Manages the files created with the File transfer protocol.
Firmware Pack Updater Handles firmware upgrade and downgrade operations.
(FPU)
Host Configuration (HOC) Configures network parameters that apply to the Aastra unit (not
to a specific interface).
Local Quality Of Service Configures the packets tagging sent from the Aastra unit.
(LQOS)
Process Control Manager Responsible to boot and restart the unit.
(PCM)
Service Controller Manager Responsible to:
(SCM) • Manage services information.
• Offer proxy functionality for service interoperation.
User Services
Basic Network Interface Configures the IP address and network mask for the Uplink and
(BNI) LAN1 networks.
Call Routing (CROUT) Routes calls between interfaces.
Call Detail Record (CDR)
Command Line Interface Allows you user to configure the unit parameters by, Telnet or
(CLI) SSH.
CPE WAN Management Not applicable.
Protocol (CWMP)
DHCP Server (Dhcp) Allows the user to lease IP addresses and send network
configuration to hosts located on any network.
Endpoint Administration Holds basic administration and status at endpoint and unit level.
(EpAdm)
Endpoint Services (EpServ) Manages endpoint behaviour and holds configuration
parameters related to endpoints (such as DTMF maps,
telephony services, etc.).
IP Routing (IpRouting) Allows the user to configure the unit's routing table.
IP Synchronization (IpSync) Controls the IP media synchronization using clock reference
signals sent over IP.

Services Table Software Configuration Guide
Table 15: Aastra unit Services (Continued)
Service Description
Integrated Services Digital Not applicable.

Network (ISDN)
Local Firewall (LFW) Allows you to filter incoming packets whose final destination is
the unit.
Link Layer Discovery Used by network devices for advertising their identity,
Protocol (Lldp): capabilities, and neighbors on a IEEE 802 local area network,
usually wired Ethernet.
Media IP Transport (MIPT) Holds basic configuration parameters (such as voice/data
codec) and implements basic functionality related to media
stream.
Music on Hold (MOH) Allows you to configure the Music on Hold parameters.
Network Address Translation Allows the user to change the source or destination address/port
(Nat) of a packet.
Network Firewall (Nfw) Allows the user to filter forwarded packets.
Notifications and Logging Handles syslog messages and notification messages.
Manager (NLM)
Network Traffic Control (Ntc) Controls the bandwidth limitation applied to physical network
interfaces.
Plain Old Telephony System Holds basic configuration parameters (such as DTMF dialing
Lines service (POTS) delays) and implements basic functionality related to POTS lines
(such as enabling/disabling individual lines).
SIP Endpoint (SipEp) Manages the behaviour of the system regarding SIP.
SNMP (SNMP) Accesses internal variables through an SNMP client. It also
handles user authentication.
Telephony Interface (TELIF) Configures the basic specification of each telephony interface.
Web (WEB) Allows accessing the unit through web pages, using HTTP.
2. In the User Service section, select the service startup type of a service in the Startup Type column.
Table 16: Startup Types
Type Description
Auto The service is automatically started when the system starts.

Manual The administrator must manually start the service.
You can put only user services in manual startup type. Proceed with caution when setting services
to manual because this could prevent you from successfully contacting the unit.
3. Select if you want to perform service commands on one or more services in the Action column.
Table 17: Actions
Action Description
Starts the service.
Stops the service.

Table 17: Actions
Action Description
Restarts the service.
When a service needs to be restarted to apply new configuration you have set elsewhere in the web
interface, it is displayed in bold and the message Restart needed is displayed in the Comment
column.
If you stop, start or restart a service, any dependent services are also affected. The tabs of the
services that have been stopped or have never been started because their startup type is manual
are greyed out. Upon clicking these tabs, a list of services that must be restarted is displayed.
4. Click the Restart Required Services button at the bottom of the page.
Graceful Restart of Services

You can set a delay to allow for telephony calls to be all completed before restarting services that need a
restart.
During that delay, it is impossible to make new calls but calls in progress are not terminated. When all calls
are completed, then the restart is authorized and the services that require a restart are restarted.
You can also set a unit restart grace period when performing a Firmware Upgrade as described in “Firmware
Packs Configuration” on page 425.
 To configure the graceful restart of services:

1. In the Restart Required Services section, set the Graceful Delay field with the delay (in minutes)
allowed for telephony calls to be all completed.
At the expiration of this delay, the services are forced to restart.
Figure 11: Services – Restart Required Services Section

1
2. Click Restart Required Services to restart only the services that needed a restart for their
configuration to be applied.
If you click Cancel, this cancels the restart during the grace delay period.
Restarting a Service via MIB

If you are using a MIB browser to access the Aastra unit configuration via SNMP, you can determine whether
or not a service needs to be restarted by locating the configurationGroup folder of the related service and
checking if the service needs to be restarted in the needRestartInfo variable.
Figure 12: Need Restart Info
If a specific service needs to be restarted, locate the scmMIB, then set the serviceCommandsRestart variable
for this service to restart.

Services Table Software Configuration Guide
Figure 13: Restart Service
You can also start a service by setting the serviceCommandsStart variable for this service to Start.
You can also stop a service by setting the serviceCommandsStop variable for this service to Stop.
If you are not able to restart a service because it is a system service, you must restart the Aastra unit.


C H A P T E R
5 Hardware Parameters
Hardware Card Configuration
For Aastra unit models that have two Ethernet ports, you can configure how each port provides a link interface.
 To configure the bridging parameter:

1. In the web interface, click the System link, then the Hardware sub-link.
Figure 14: System – Hardware Web Page (Aastra TA7102i shown)
2. In the Unit Configuration section, set the Eth Ports drop-down menu with the proper behaviour.
Table 18: Bridging Parameters
Separate Each Ethernet port provides an independent link interface. This is the required
configuration for IP Routing.
Bridge Both Ethernet ports are bridged together and provide a single link interface.
3. Click Submit if you do not need to set other parameters.

The following message displays:
Note: Your Ethernet configuration has changed. A link interface will be deactivated. Make sure that
your network interfaces are configured accordingly prior to restarting the unit.
4. In the web interface, click the Network link, then the Interfaces sub-link.
Figure 15: Network – Interfaces Web Page
6 7 8 9 10
5. Enter the name of the new interface for bridging in the blank field in the bottom left of the window,
then click the button.
The name is case-sensitive. Using the special value “All” is not allowed.

Chapter 5 - Hardware Parameters Hardware Card Configuration
6. In the Interface Configuration section, select the link on which to activate the interface in the Link
column.
Select the link associated with the bridge. The name varies depending on the platform used.
7. Select the configuration source of the interface information in the Type drop-down menu.
Table 19: Interface Configuration Sources
Source Description
IPv4 The IPv4 address and network mask are provided by querying a DHCP server and
DHCP using standard DHCP fields or options. Using the DHCP configuration assumes that
you have properly set your DHCP server with the relevant information. DHCP servers
may provide a list of IP configuration parameters to use. See “DHCP Server
IPv4 You manually enter the IPv4 address and network mask and they remain the same
Static every time the Aastra unit restarts. Use the static configuration if you are not using a
DHCP server/PPP peer or if you want to bypass it.
IPv4 IPv4 over PPP connection, address and network mask are provided by the PPP peer
PPPoE using IPCP. PPP peers may provide a list of IP configuration parameters to use. See
“PPPoE Configuration” on page 91 for more details.
IPv6 IPv6 state-less auto-configuration.
Auto-
Conf
Static every time the Aastra unit restarts. Use the IPv6 static configuration if you are not using
IPv6 stateless or stateful auto-configuration or if you want to bypass it.
Note: If no network is configured in IPv6, the unit does not have any IPv6 address, not even the Link-Local
address. When a network is configured in IPv6, the Link-Local (FE80 ::...) address is automatically created
and displayed in the Network Status information.
8. If the interface configuration source is IPv4 Static or IPv6 Static, enter the address and network
mask (if applicable) of the network interface in the Static IP address field.
9. If the interface configuration source is IPv4 Static or IPv6 Static, set the Static Default Router field
with the IP address of the default gateway for the network interface.
10. Define whether or not the Aastra unit should attempt to activate the corresponding network interface
in the Activation drop-down menu.
It may not be possible to enable a network interface, for instance if another network interface is
already enabled in the same subnet. The actual status of network interfaces is shown in the Status
page.
Note: The newly created interface will be the only valid interface after the restart, make sure this interface
is Enabled and correctly configured according on the Interface Configuration Source (your network).

The current network interface information is displayed in the Status page. See “Interfaces
Configuration” on page 100 for more details on network interfaces.
12. Restart the Aastra unit to apply the change.
Ring Management

Hardware Card Configuration Software Configuration Guide

You can determine how to ring more than one port. You have the following choices:
Table 20: Ring Management Parameters
Parameter Descriprion
Cascade The FXS ports are prevented from ringing at the same time in order to reduce the peak
power usage of the device.
Simultaneous All ports are ringing at the same time.
 To set the ring management:

1. In the MbLdpMIB, set the RingManagement variable to the proper value.
MbLdp.RingManagement="Value"

Table 21: Ring Management Values
Value Description
100 Cascade
200 Simultaneous

Chapter 5 - Hardware Parameters Hardware Card Configuration

C H A P T E R
6 Endpoints State Configuration
This chapter describes how to set the administrative state of the Aastra unit’s endpoints.
Unit Configuration
The unit configuration section allows you to define the administrative state of all the Aastra unit’s endpoints.
 To set the unit’s endpoints parameters:

1. In the web interface, click the System link, then the Endpoints sub-link.
Figure 16: System Configuration – Endpoints Web Page
2. In the Unit States section, select a temporary state for all of the unit’s endpoints in the Action
column.
This command locks/unlocks all endpoints of the Aastra unit. This state is kept until you modify it or
the unit restarts. It offers the following settings:
Table 22: Action Settings
Setting Description
Force Lock Cancels all the endpoints registration to the SIP server. All active calls in
progress are terminated immediately. No new calls may be initiated.
Lock Cancels all the endpoints registration to the SIP server. Active calls in
progress remain established until normal call termination. No new calls
may be initiated.
Unlock Registers the endpoints to the SIP server.
3. If you do not need to set other parameters, click Submit.
Endpoints Configuration
The endpoints configuration allows you to define the administrative state of the Aastra unit’s endpoints.

Chapter 6 - Endpoints State Configuration Administration
 To set the endpoints parameters:

1. In the Endpoint States section of the Endpoints page, select the permanent administrative state
each endpoint will have when the Aastra unit restarts in the Initial Administrative column.
Figure 17: Endpoint States Section

1 2
Table 23: Permanent Administrative State Settings
Setting Description
Unlocked Registers the endpoint to the SIP server.

Locked The endpoint is unavailable for normal operation. It cannot be used to
make and/or receive calls.
2. Select a temporary state for each endpoint in the corresponding Action column.
This command locks/unlocks an endpoint of the Aastra unit. This state is kept until you modify it or
the unit restarts. It offers the following settings:
Table 24: Action Settings
Setting Description
Force Lock Cancels the endpoint registration to the SIP server. All active calls in
progress are terminated immediately. No new calls may be initiated.
Lock Cancels the endpoint registration to the SIP server. Active calls in
progress remain established until normal call termination. No new calls
may be initiated.
Unlock Registers the endpoint to the SIP server.
3. If you do not need to set other parameters, click Submit.
Administration
The Administration section allows you to define endpoint operational state.
 To set administration parameters:

1. In the Administration section of the Endpoints page, set the Disable Unit (All Endpoints) When No
Gateways Are In State Ready drop-down menu with the proper behaviour.
Figure 18: Administration Section
1
2
Table 25: Unit Operational State Parameters
Disable Signaling gateways have no impact on the unit operational

state

Administration Software Configuration Guide
Table 25: Unit Operational State Parameters (Continued)
Enable When all signaling gateways are not ready, the unit
operational state is set to disabled.
2. Set the Shutdown Endpoint When Operational State is Disable And Its Usage State Is 'idle-
unusable' drop-down menu with the proper behaviour.
Table 26: Endpoint Shutdown Parameters
Enable When the usage state becomes “Idle-unusable” and the

operational state becomes “Disable”, the endpoint is
physically shutdown.
Disable When an endpoint's usage state becomes “Idle-unusable”
whatever the value of its operational state, the endpoint
remains physically up but the calls are denied.
The default value is:

• Enable for the Aastra series
Unit Shutting Down Behaviour

You can configure the behaviour of the call permissions when the UnitAdminState is ShuttingDown.
The following parameters are available:
Table 27: Unit Shutting Down Behaviour Parameters
BlockNewCalls No new requests are accepted once all activity are terminated. Endpoints cannot make and
receive calls.
AllowNewCalls New requests are accepted until all activities are simultaneously terminated. Endpoints can
make and receive calls.
 To set the unit shutting down behaviour:

1. In the epAdmMIB, locate the UnitConfigGroup folder.
2. Set the behaviorWhileInUnitShuttingDownState variable with the proper behaviour.
epAdm.behaviorWhileInUnitShuttingDownState="Value"
where Value may be one of the following:
Table 28: Unit Shutting Down Behaviour Values
Value Meaning
100 BlockNewCalls

Chapter 6 - Endpoints State Configuration Administration
Table 28: Unit Shutting Down Behaviour Values (Continued)
Value Meaning
200 AllowNewCalls

C H A P T E R
7 Syslog Configuration
This chapter describes how the Aastra unit. handles syslog messages and notification messages.
For a list and description of all syslog messages and notification messages that the Aastra unit may send, refer
to the Notification Reference Guide.
Syslog Daemon Configuration
Standards Supported • RFC 3164: The BSD Syslog Protocol
The Syslog daemon is a general purpose utility for monitoring applications and network devices with the TCP/
IP protocol. With this software, you can monitor useful messages coming from the Aastra unit. If no Syslog
daemon address is provided by a DHCP server or specified by the administrator, no messages are sent.
For instance, if you want to download a new firmware into the Aastra unit, you can monitor each step of the
firmware download phase. Furthermore, if the unit encounters an abnormal behaviour, you may see accurate
messages that will help you troubleshoot the problem.
The Aastra unit supports RFC 3164 as a “device” only (see definition of device in section 3 of the RFC).

Chapter 7 - Syslog Configuration Syslog Daemon Configuration
 To configure the Aastra unit syslog client:

1. In the web interface, click the System link, then the Syslog sub-link.
Figure 19: System – Syslog Web Page
4
5
2. Set the static IP address or domain name and port number of the device to use to archive log entries
in the Remote Host field.
Use the special port value zero to indicate the protocol default. For instance, the TFTP default port
is 69 and the HTTP/HTTPS default port is 80.
3. In the Service Severity section, select the minimal severity to issue a notification message for the
various services in the corresponding drop-down menus.
Any syslog message with a severity value greater than the selected value is ignored. Available
values are:
Table 29: Severity Values
Severity Description Notification Messages Issued
Disable N/A No notification is issued.

Debug Message describing in detail the unit's All notification messages are issued.
operations.
Info Message indicating a significant event for Notification messages with severity
the unit's normal operations. “Informational” and higher are issued.
Warning Message indicating an abnormal event or Notification messages with severity
situation that could be potentially risky. “Warning” and higher are issued.
The unit may not be fully operational.

Syslog Daemon Configuration Software Configuration Guide
Table 29: Severity Values (Continued)
Severity Description Notification Messages Issued
Error Message indicating an abnormal event or Notification messages with severity

situation, the system's operation is “Error” and higher are issued.
affected. The unit may not be operational.
Critical Message indicating a critical event or Notification messages with severity
situation that requires immediate “Critical” are issued.
attention. The unit is not operational.
A higher level mask includes lower level masks, e.g., Warning includes Error and Critical. The
default value is Warning.
4. In the Technical Assistance Centre section, enable diagnostic traces by setting the Diagnostic
Traces drop-down menu to Enable.
At the request of Aastra’s Technical Support personnel, enabling these traces will allow Aastra to
further assist you in resolving some issues. However, be advised that enabling this feature issues
a lot of messages to the syslog host. These messages may be filtered by using the Diagnostic
Traces Filter field.
Note: Enabling all the traces could affect the performance of the Aastra unit.
5. If applicable, define the filter applied to diagnostic traces by clicking the Edit button in the Filter field.
The following opens:
Figure 20: Diagnostic Traces Window
You can use the filter to narrow down the number of traces sent at the request of Aastra´s Technical
Support personnel.

Configuring PCM Capture

Aastra Technical Support personnel may ask you to enable the PCM traces. PCM traces are an efficient tool
to identify problems with:
 Echo in your network
 DTMF signals
 Caller ID signals
 Fax signals (or false Fax detection)
 Message Waiting Indicator signals
 Any other analog or digital signal
The PCM traces are two different RTP streams made specifically to record all analog signals that are either
sent or received on the analog side of the Aastra unit. Only the configured port, port #1 and/or #2, are sending
the PCM traces for a maximum of four simultaneous RTP streams.
The RTP streams are sent to a configurable IP address, normally an IP address on your network where it can
be recorded with a packet sniffer (such as Wireshark). Moreover, they are independent from the regular RTP
streams of the VoIP call.
PCM capture supports sending streams with a ptime higher than 20 ms. The Aastra unit does not support
sending streams with a ptime of 10 ms.
 To enable PCM capture:

1. In the miptMIB, locate the pcmCaptureGroup folder under the debugGroup folder.
2. Set the pcmCaptureEnable variable to enable.
mipt.pcmCaptureEnable="1"
3. Set the unit's endpoint on which the PCM capture must be performed in the pcmCaptureEndpoint
variable.
mipt.pcmCaptureEndpoint="Value"
The format is InterfaceName-Channel#. For digital interfaces (such as ISDN), you must append a
-Channel# for the requested channel.
The list of endpoints is available under EpAdm.EndpointTable. Valid examples (depending of the
platform) are:
• PCM capture is to be done on channel #3 of a PRI interface located in slot #2: Slot2/
E1T1-3
• PCM capture is to be done on channel #2 of a BRI interface: Bri1-2
• PCM capture is to be done on the 16th FXS port: Port16
Note: Note that PCM capture does not support capturing on multiple endpoints simultaneously.
4. Set the IP address where the captured PCM packets should be sent in the pcmCaptureIpAddr
variable.
mipt.pcmCaptureIpAddr="Value"

Syslog Daemon Configuration Software Configuration Guide
The PCM traces destination must be set so it can be recorded in a Wireshark capture on your
network, normally sent to the PC doing the capture.
Configuring the Syslog Daemon Application

You must configure the Syslog daemon server to capture those messages. Refer to your Syslog daemon’s
documentation to learn how to properly configure it to capture messages.


C H A P T E R
8 Events Configuration
This chapter describes how to associate a NOTIFICATION message and how to send it (via syslog or via a
SIP NOTIFY packet).
For a list and description of all syslog messages and notification messages that the Aastra unit may send, refer
to the Notification Reference Guide.
Notification Events
You can configure an event router in order to apply a set of rules to select the proper transport protocol
scheme. A rule entry is made up of three different values: type, criteria and action.
Note that more than one notification may be sent for a single event based on the event router table rules.
 To configure notification events:

1. Ensure that the severity level for all services are set according to the severity level of the notification
messages that are required by the system administrator. See “Chapter 7 - Syslog Configuration” on
2. In the web interface, click the System link, then the Events sub-link.
Figure 21: System – Events Web Page
4 5 6 7 8
3. If you want to add a rule entry before an existing entry, locate the proper row in the table and click
the button of this row.
4. Set the Activation drop-down menu with the current activation state for the corresponding system
event.
Table 30: Activation Parameters
Enable This action is enabled for this system event.

Disable This action is disabled for this system event.
5. Optional: Set the corresponding Criteria field with the expression an event must match in order to
apply the specified action. The expression is based on the event type.
This step is optional because a proper value may be automatically entered by the Aastra unit upon
setting the Service (Step 5) and Notification (Step 6) drop-down menus.

Chapter 8 - Events Configuration Notification Events
An event of type notification uses the notification ID as expression criteria. The notification ID is the
combination of the service number key and the message number key separated by a dot. The
information regarding the service and message number key is available in the Notification
Reference Guide document.
Several basic criteria can also be specified on the same line, separated by commas. Criteria can
specify inclusion or exclusion. A group of exclusion criteria can follow the group of inclusion criteria.
The group of exclusion criteria must begin with a hyphen (-).
Matching an inclusion criteria causes the action to be executed unless an exclusion criteria is also
matched. Exclusion criteria have precedence over inclusion criteria.
Spaces are allowed before or after a basic criterion; however, spaces are not accepted within a
basic criterion, i.e. before or after the dot.
Examples:
Service ISDN (number key = 1850)
Message %1$s: Physical link state changed to up (number key = 5)
The corresponding Criteria is: 1850.5
You can also use the special expression All, which means all available services and messages.
Criteria 1850.All,1600.200,1600.W,-1850.500,1600.300
1850.All,1600.200,1600.W are inclusion criteria and -1850.500,1600.300 are exclusion criteria. All
notifications from service 1850, except notification 500, will match the expression. All notifications
from service 1600 with Warning level, except notification 300, will match the expression. Notification
200 from service 1600 will match the expression, no matter the severity level.
6. In the corresponding Service drop-down menu, select the service for which you want to send
events.
You can use the Suggestion column’s drop-down menu to select between suggested values, if any.
7. In the Notification drop-down menu, select the notification message that you want to send.
8. In the Action drop-down menu, select the action to apply to the system event if the criteria matches.
The action represents a transport targeted for the event. The format of the event under which the
message is carried is dependent on the protocol in use.
The possible actions are:
Table 31: Action Parameters
Send Via The event notification is sent using syslog as transport. See “Chapter 7 - Syslog
Syslog Configuration” on page 71 for more details.
Send Via The event notification is sent using SIP Notify as transport.
SIP
Log The event notification is logged in Local Log.
Locally
9. Click the Submit button.

The configuration status of the row displays on the right part of the row. It indicates whether the
configuration of the row is valid.
Table 32: Configuration Status Values
Value Description
Valid The current content of the fields Type, Criteria and Action is valid.
Invalid The current content of the fields Type, Criteria and Action is not valid.

Notification Events Software Configuration Guide
Table 32: Configuration Status Values (Continued)
Value Description
Not The current content of the fields Type, Criteria and Action is valid but not
Supported supported.
Deleting a Rule
You can delete a rule row from the table in the web interface.
 To delete a rule entry:

1. Click the button of the row you want to delete.
Monitoring Parameters
You can set two monitoring parameters for the Notification Events table.
 To set monitoring parameters:

1. In the sipEpMIB, locate the MonitoringGroup folder.
2. Set the sipNotificationsGateway variable with the SIP gateway used to send SIP NOTIFY
containing the notification events.
sipEp.sipNotificationsGateway="Value"
Value is the name of the SIP gateway from which the NOTIFICATION is sent.
3. Set the maxNotificationsPerNotify variable with the maximal number of notification events the
device may have to send in one SIP NOTIFY request.
Notifications are sent in XML elements through the SIP NOTIFY's body request.
sipEp.maxNotificationsPerNotify="Value"
Value may be between 1 and 25.

Chapter 8 - Events Configuration Notification Events

C H A P T E R
9 Local Log
This chapter describes local log status and entries for your Aastra unit.
Local Log Status and Entries
You can display, clear and refresh local log status and entries.
 To manage local log status and entries:

1. In the web interface, click the System link, then the Local Log sub-link.
Figure 22: System – Local Log Web Page
2 3
The following is the Local Log Status information displayed.

Table 33: Local Log Status Parameters
Maximum Number of Entries Maximum number of entries that the local log can contain. When
adding a new entry while the local log is full, the oldest entry is
erased to make room for the new one.
Number of Error Entries Current number of error entries in the local log.
Number of Critical Entries Current number of critical entries in the local log.
The following is the Local Log Entries information displayed.

Table 34: Local Log Entries Parameters
Local Time Local date and time at which the log entry was inserted. Format is
YYYY-MM-DD HH:MM:SS.
Severity Severity of the log entry.
Service Name Textual identifier of the service that issued the log entry.
Service Key Numerical identifier of the service that issued the log entry.
Message Key Numerical identifier of the notification message.
Message Content The readable content of the log message.

Chapter 9 - Local Log Local Log Status and Entries
2. Click Clear Local Log to clear all log entries.

3. Click Refresh Local Log to refresh the log entries display.

Network Parameters
C H A P T E R
10 IPv4 vs. IPv6
This chapter describes the differences between IPv4 and IPv6 addressing.
Introduction
IPv6 (Internet Protocol version 6) is the successor to the most common Internet Protocol today (IPv4). This is
largely driven by the fact that IPv4’s 32-bit address is quickly being consumed by the ever-expanding sites and
products on the internet. IPv6’s 128-bit address space should not have this problem for the foreseeable future.
IPv6 addresses, in addition to being longer, are distinguished from IPv4 addresses by the use of colons ":",
e.g., 2001:470:8929:4000:201:80ff:fe3c:642f. An IPv4 address is noted by 4 sets of decimal numbers
separated by periods ".", e.g., 192.168.10.1.
Please note that IPv6 addresses should be written between [ ] to allow port numbers to be set. For instance:
[fd0f:8b72:5::1]:5060.
IPv4 vs. IPv6 Availability
The Aastra unit fully supports IPv4 IP addresses, as well as IPv6 IP addresses in some of its features. The
following table lists all the network related features of the Aastra unit with their availability in IPv4 and IPv6.
Table 35: IPv4 vs. IPv6 Availability
Feature IPv4 IPv6
Backup/Restore transfer
Command Line Interface (CLI)
Configuration file transfer
Embedded DHCP server
Firmware Transfer
IP Routing
IP Sync
Link Layer Discovery Protocol (LLDP) QoS settings
Local Firewall (LFW)
Network Address Translation (NAT)
Network Configuration (IP addresses, DNS and SNTP servers)
Network Firewall (NFW)
Online Certificate Status Protocol (OCSP)
Remote Authentication Dial In User Service (Radius )
SIP signaling and media transport
Simple Network Management Protocol (SNMP)

Chapter 10 - IPv4 vs. IPv6 IPv6 Scope Identifier
Table 35: IPv4 vs. IPv6 Availability (Continued)
Feature IPv4 IPv6
TR-069
WEB Configuration
If you configure the Aastra unit with IPv6 addresses, then decide to go downgrade to a firmware version that
does not support IPv6, all IPv6 networks are deleted.
Please note that IPv6 addresses should be written between [ ]. For instance: [fd0f:8b72:5::1].
IPv6 Scope Identifier
When using an IPv6 address starting with "FE80::" (IPv6 link-local addresses), there must be additional
information: the IPv6 scope identifier (this represents the network link that will be used to contact the IPv6 link-
local address). The format is "[IPv6 link-local%ScopeIdentifier]".
When Contacting the unit using its IPv6 link-local Address

On Windows, the scope identifier is represented by an interface number. The interface number can be
determined through the command line of Windows.
 Go to Start -> Run and type cmd to enter the command prompt.
 At the command prompt, type ipconfig and find the IPv6 address. Appended to the end of this
will be a "%x" where x is the interface number.
To contact the IPv6 link-local IPv6 address "fe80::201:80ff:fe3c:642f", you would use:
[fe80::201:80ff:fe3c:642f%4]
On Linux, the scope identifier may be the link name or the interface number. The interface number can be
determined through the Linux command line.
To contact the IPv6 link-local IPv6 address "fe80::201:80ff:fe3c:642f", you would use:
[fe80::201:80ff:fe3c:642f%2] or [fe80::201:80ff:fe3c:642f%eth0]

IPv6 Scope Identifier Software Configuration Guide
When Configuring the Aastra unit to use an IPv6 link-local Address

In that case, the scope identifier represents the "link" in Network/Interfaces.
For instance, if you want your unit to contact a server with the address IPv6 link-local
"fe80::201:80ff:fe3c:642f", you must check on which network link the server is available. Some units have
"wan" or "lan". Let’s say it is on the "wan" link. The IP address whoud then become
"[fe80::201:80ff:fe3c:642f%wan]".

Chapter 10 - IPv4 vs. IPv6 IPv6 Scope Identifier

C H A P T E R
11 Host Parameters
This chapter describes how to set the host information of the Aastra unit:
 General Configuration (automatic configuration interface)
 Host name and domain name.
 Default gateway parameters.
 DNS parameters.
 SNTP client parameters.
 Time parameters.
General Configuration
The General Configuration section allows you to configure the networks that will provide the automatic
configuration (host name, default gateway, DNS servers and SNTP servers) used by the Aastra unit.
Automatic configuration may be provided via IPv4 (DHCPv4) and/or via IPv6 (stateless auto-configuration and
DHCPv6).
 To set the general configuration:

1. In the web interface, click the Network link, then the Host sub-link.
Figure 23: Network – Host Web Page
2
3
2. Set the Automatic IPv4 config source network drop-down menu with the IPv4 network interface that
provides the automatic configuration.
3. Set the Automatic IPv6 config source network drop-down menu with the IPv6 network interface that
provides the automatic configuration.
The current automatic configuration interface is displayed in the Status page.
Host Configuration
The Host Configuration section allows you to configure the host name and domain name of the Aastra unit.

Chapter 11 - Host Parameters Host Configuration
 To set the host configuration:

1. In the Host Configuration section of the Host page, select the configuration source of the domain
name information in the Domain Name Configuration Source drop-down menu.
Figure 24: Host Name Configuration Section
2
3
Table 36: Host Name Configuration Sources
Source Description
Automatic The domain name is automatically obtained from the network. The value obtained
IPv4 depends on the connection type of the automatic network interface (see “General
Configuration” on page 75) if any. Using the automatic configuration assumes that
you have properly set your network server with the relevant information.
Note: Some Uplink connection types (for example Static and PPPoE) cannot obtain
domain name information from the network, and therefore lead to no domain name
being applied to the system.
Automatic The domain name is automatically obtained from the IPv6 network defined in the
IPv6 Automatic IPv6 config source network drop-down menu.
Static You manually enter the domain name and it remains the same every time the Aastra
unit restarts. Use the static configuration if you are not using a network server or if
you want to bypass it.
When switching from the Static to Automatic IPv4 or Automatic IPv6 configuration source, the last
value correctly obtained from the network (if any) is applied to the system.
Static Configuration Source Only
2. Set the system’s domain name in the Domain Name field.

A domain name is a name of a device on the Internet that distinguishes it from the other systems
on the network. For instance: example.com.
3. Set the system's host name in the Host Name field.
The host name is the unique name by which the device is known on a network. It may contain any
of the following characters:
• A to Z and a to z letters
• 0 to 9 digits
• -._~
• !$&'()*+=
Certain restrictions apply to this name:
• The host name must be shorter than 64 characters.
• The host name must not start with a period.
• The host name must not contain double quotes, semicolons, curly braces, spaces, and
commas.
• The host name must not contain the following characters: :/?#[@
The current domain name is displayed in the Status page.

Default Gateway Configuration Software Configuration Guide
Default Gateway Configuration
The default gateway (also known as default router) is the gateway to which the Aastra unit sends packets
when all other internally known routes have failed.
 To set the default gateway configuration:

IPv4 Configuration
1. In the Default Gateway Configuration – IPv4 section of the Host page, select the IPv4 configuration
source of the default gateway information in the Configuration Source drop-down menu.
Figure 25: Default Gateway Configuration Section
1
2
3
4
Table 37: Default Gateway Configuration Sources
Source Description
Automatic The default gateway is automatically obtained from the network. The value obtained
Note: Some Uplink connection types (for example Static) cannot obtain default
gateway information from the network, and therefore lead to no default gateway being
applied to the system.
Static You manually enter the IP address of the default gateway and it remains the same
every time the Aastra unit restarts. Use the static configuration if you are not using a
network server or if you want to bypass it.
When switching from the Static to Automatic configuration source, the last value correctly obtained
from the network (if any) is applied to the system.
IPv4 Static Configuration Source Only
2. If the default gateway configuration source is Static, enter the static default gateway address in the
IP address field.
This can be an IP address or domain name. The default value is 192.168.10.10.
IPv6 Configuration
3. In the Default Gateway Configuration – IPv6 section of the Host page, select the IPv6 configuration
source of the default gateway information in the Configuration Source drop-down menu.
Table 38: IPv6 Default Gateway Configuration Sources
Source Description
Automatic The default gateway name is automatically obtained from the IPv6 network defined
IPv6 in the Automatic IPv6 config source network drop-down menu.
Static You manually enter the IPv6 address of the default gateway and it remains the same

Chapter 11 - Host Parameters DNS Configuration
When switching from the Static to Automatic IPv6 configuration source, the last value correctly
obtained from the network (if any) is applied to the system.
4. If the default gateway configuration source is Static, enter the static default gateway IPv6 address
in the IP address field.
This can be an IP address or domain name.
The current default gateway address is displayed in the Status page.
DNS Configuration
Standards Supported • RFC 1034: Domain Names - Concepts and Facilities

• RFC 1035: Domain Names - Implementation and
Specification
• RFC 1886: DNS Extensions to support IP version 6
• RFC 2181: Clarifications to the DNS Specification
You can use up to four Domain Name Servers (DNS) to which the Aastra unit can connect. The DNS servers
list is the ordered list of DNS servers that the Aastra unit uses to resolve network names. DNS query results
are cached on the system to optimize name resolution time.
 To set the DNS configuration:

1. In the DNS Configuration section of the Host page, select the configuration source of the DNS
information in the Configuration Source drop-down menu.
Figure 26: DNS Configuration Section
Table 39: DNS Configuration Sources
Source Description
Automatic The DNS servers are automatically obtained from the network. The value obtained
Note: Some Uplink connection types (for example Static) cannot obtain DNS
information from the network, and therefore lead to no DNS servers being applied to
the system.
Automatic The DNS servers are automatically obtained from the IPv6 network defined in the
IPv6 Automatic IPv6 config source network drop-down menu.
Static You manually enter up to four DNS servers IP addresses and they remain the same
values correctly obtained from the network (if any) are applied to the system.

SNTP Configuration Software Configuration Guide
2. If the DNS configuration source is Static, enter up to four static DNS addresses in the following
fields:
• Primary DNS
• Secondary DNS
• Third DNS
• Fourth DNS
The current list of DNS servers is displayed in the Status page.
SNTP Configuration
Standards Supported • RFC 2030: Simple Network Time Protocol (SNTP) Version 4
for IPv4, IPv6 and OSI
• bootp-dhcp-option-88
The Simple Network Time Protocol (SNTP) enables the notion of time (date, month, time) into the Aastra unit.
SNTP is used to synchronize a SNTP client with a SNTP or NTP server by using UDP as transport. It updates
the internal clock of the unit to maintain the system time accurate. It is required when dealing with features
such as the caller ID.
The Aastra unit implements a SNTP version 3 client.
Note: The Aastra unit hardware does not include a real time clock. The unit uses the SNTP client to get and
set its clock. As certain services need correct time to work properly (such as HTTPS), you should configure
your SNTP client with an available SNTP server in order to update and synchronise the local clock at boot
time.
 To set the SNTP client of the Aastra unit:

1. In the SNTP Configuration section of the Host page, select the configuration source of the SNTP
Figure 27: SNTP Configuration Section
Table 40: SNTP Configuration Sources
Source Description
Automatic The SNTP parameters are automatically obtained from the network. The value
IPv4 obtained depends on the connection type of the automatic network interface (see
“General Configuration” on page 75) if any. Using the automatic configuration
assumes that you have properly set your network server with the relevant
information.
Note: Some Uplink connection types (for example Static and PPPoE) cannot obtain
SNTP information from the network, and therefore lead to no SNTP parameters being
applied to the system.
Automatic The SNTP parameters are automatically obtained from the IPv6 network defined in
IPv6 the Automatic IPv6 config source network drop-down menu.

Chapter 11 - Host Parameters Time Configuration
Table 40: SNTP Configuration Sources (Continued)
Source Description
Static You manually enter the values and they remain the same every time the Aastra unit
restarts. Use the static configuration if you are not using a network server or if you
want to bypass it.
values correctly obtained from the network (if any) are applied to the system.
2. If the SNTP configuration source is Static, enter up to four static SNTP server IP addresses or
domain names and port numbers in the following fields:
• Primary SNTP
• Secondary SNTP
• Third SNTP
• Fourth SNTP
3. Set the synchronization information:
Table 41: SNTP Synchronization Information
Field Description
Synchronisation Period Time interval (in minutes) between system time

synchronization cycles. Each time this interval expires, a
SNTP request is sent to the SNTP server and the result is
used to set the system time. The maximum value is set to
1 440 minutes, which corresponds to 24 hours.
Synchronisation Period on Error Time interval (in minutes) between retries after an
unsuccessful attempt to reach the SNTP server. The
maximum value is set to 1 440 minutes, which corresponds to
24 hours.

The current SNTP host is displayed in the Status page.
Time Configuration
Standards Supported • bootp-dhcp-option-88
You can define the current system date and time configured in the unit by specifying in which time zone the
unit is located.
If the time seems not valid, verify the SNTP configuration in “SNTP Configuration” on page 79.
 To set the time of the Aastra unit:

1. In the Time Configuration section of the Host page, enter a valid string in the Static Time Zone field.
Figure 28: Time Configuration Section
The format of the string is validated upon entry. Invalid entries are refused. The default value is:
EST5DST4,M4.1.0/02:00:00,M10.5.0/02:00:00

Time Configuration Software Configuration Guide
A POSIX string is a set of standard operating system interfaces based on the UNIX operating
system. The format of the IEEE 1003.1 POSIX string is defined in the bootp-dhcp-option-88 Internet
draft as:
STDOFFSET[DST[OFFSET],[START[/TIME],END[/TIME]]]
Refer to the following sub-sections for explanations on each part of the string.
The current system time is displayed in the Status page.
STD / DST
Three or more characters for the standard (STD) or alternative daylight saving time (DST) time zone. Only STD
is mandatory. If DST is not supplied, the daylight saving time does not apply. Lower and upper case letters are
allowed. All characters are allowed except digits, leading colon (:), comma (,), minus (-), plus (+), and ASCII
NUL.
OFFSET
Difference between the GMT time and the local time. The offset has the format h[h][:m[m][:s[s]]]. If no offset is
supplied for DST, the alternative time is assumed to be one hour ahead of standard time. One or more digits
can be used; the value is always interpreted as a decimal number.
The hour value must be between 0 and 24. The minutes and seconds values, if present, must be between 0
and 59. If preceded by a minus sign (-), the time zone is east of the prime meridian, otherwise it is west, which
can be indicated by the preceding plus sign (+). For example, New York time is GMT 5.
START / END
Indicates when to change to and return from the daylight saving time. The START argument is the date when
the change from the standard to the daylight save time occurs; END is the date for changing back. If START
and END are not specified, the default is the US Daylight saving time start and end dates. The format for start
and end must be one of the following:
 n where n is the number of days since the start of the year from 0 to 365. It must contain the
leap year day if the current year is a leap year. With this format, you are responsible to
determine all the leap year details.
 Jn where n is the Julian day number of the year from 1 to 365. Leap days are not counted. That
is, in all years – including leap years – February 28 is day 59 and March 1 is day 60. It is
impossible to refer to the occasional February 29 explicitly. The TIME parameter has the same
format as OFFSET but there can be no leading minus (-) or plus (+) sign. If TIME is not
specified, the default is 02:00:00.
 Mx[x].y.z where x is the month, y is a week count (in which the z day exists) and z is the day
of the week starting at 0 (Sunday). For instance:
M10.4.0
is the fourth Sunday of October. It does not matter if the Sunday is in the 4th or 5th week.
M10.5.0
is the last Sunday of October (5 indicates the last z day). It does not matter if the Sunday is in the
4th or 5th week.
M10.1.6
is the first week with a Saturday (thus the first Saturday). It does not matter if the Saturday is in the
first or second week.
The TIME parameter has the same format as OFFSET but there can be no leading minus (-) or plus
(+) sign. If TIME is not specified, the default is 02:00:00.

Chapter 11 - Host Parameters Additional Parameters
Example
The following is an example of a proper POSIX string:
Standard Offset Month, Week, and Day Month, Week, and Day
time zone to start the Daylight to stop the Daylight
Saving Time Saving Time
EST5DST4,M4.0.0/02:00:00,M10.5.0/02:00:00
Daylight Offset Time to start the Time to stop

Saving Time Daylight Saving the Daylight
time zone Time Saving Time
The following are some valid POSIX strings:

Table 42: Valid POSIX Strings
Time Zone POSIX String
Pacific Time (Canada & US) PST8PDT7,M3.2.0/02:00:00,M11.1.0/02:00:00

Mountain Time (Canada & US) MST7MDT6,M3.2.0/02:00:00,M11.1.0/02:00:00
Central Time (Canada & US) CST6CDT5,M3.2.0/02:00:00,M11.1.0/02:00:00
Eastern Time Canada & US) EST5EDT4,M3.2.0/02:00:00,M11.1.0/02:00:00
Atlantic Time (Canada) AST4ADT3,M3.2.0/02:00:00,M11.1.0/02:00:00
GMT Standard Time GMT0DMT-1,M3.5.0/01:00:00,M10.5.0/02:00:00
W. Europe Standard Time WEST-1DWEST-2,M3.5.0/02:00:00,M10.5.0/03:00:00
China Standard Time CST-8
Tokyo Standard Time TST-9
Central Australia Standard Time CAUST-9:30DCAUST-10:30,M10.5.0/02:00:00,M3.5.0/02:00:00
Australia Eastern Standard Time AUSEST-10AUSDST-11,M10.5.0/02:00:00,M3.5.0/02:00:00
UTC (Coordinated Universal Time) UTC0
Additional Parameters
Configuring DNS Records Randomization

You can define how the DNS A/AAAA records are accessed from the device’s internal DNS cache using the
DnsCacheRecordsRandomization variable.

Additional Parameters Software Configuration Guide
The following values are available:

Table 43: DNS Cache Records Randomization Values
Value Description
Enable When DNS A/AAAA records are accessed from the cache, they are sent to requesting
service in a randomized order.
Disable When DNS A/AAAA records are accessed from the cache, they are sent to requesting
service in the same order they were originally received from the network. This is the default
value.
 To configure DNS Cache records randomization:

1. In the hocMIB, set the DnsCacheRecordsRandomization variable.
hoc.DnsCacheRecordsRandomization=”Value”
Table 44: DNS Cache Records Randomization Values
Value Meaning
0 Disable
1 Enable
Configuring Pre-resolved Static FQDNs

You can configure up to 10 pre-resolved FQDNs. The StaticHosts table allows configuring FQDNs with static
IP addresses. When a device attempts to reach a FQDN configured in this table, the static IP addresses will
be used instead of resolving the FQDN.
Table 45: Static Host Command Parameters
Name Name (FQDN) of the static host. This name must be unique across the table.
The name only accepts valid FQDNs as defined by RFC 3986 (Uniform Resource Identifier
(URI): Generic Syntax). In addition, strict validation is applied, i.e. the suggested syntax
defined in RFC 1035 is enforced.
IpAddresses List of static IP addresses associated with the FQDN specified in the StaticHosts.Name
variable.
This list contains numerical IPv4 or IPv6 addresses. IP addresses MUST be separated by
a comma (,).
Index Index in the table. A value of zero (default) causes automatic selection of the largest
current index value + 1. If the index value already exists in the table, the insertion is
refused. This parameter is optional.
 To insert a new static host:

1. You can use one of the following lines in the CLI or a configuration script:
hoc.InsertStaticHost Index=”value” Name="hostname" IpAddresses=”address,address1”
hoc.InsertStaticHost Name="hostname" IpAddresses=”address,address1”
where:
• value can be an integer. This is an optional parameter.
• hostname is a unique valid FQDN as define by RFC 3986.

Chapter 11 - Host Parameters Additional Parameters
• address and address1 are numerical IPv4 or IPv6 addresses separated by a comma.
 To delete a static host:

1. In the hocMIB, delete the host name using the Delete command.
You can also use one of the following lines in the CLI or a configuration script:
hoc.StaticHosts.Delete[Index=value]=Delete
where value can be an integer.
Updating the "sysname" or "syslocation"

You can specify the name and location of the Aastra unit. This information is for display purposes only and
does not affect the behavior of the unit.
 To set the sysname and syslocation parameters:

1. In the hocMIB, set the system name in the systemName variable.
hoc.systemName="Value"
The value of this variable is also returned by the "sysName" object in SNMPv2-MIB.
2. Set the system location in the systemLocation variable.
hoc.systemLocation="Value"
The value of this variable is also returned by the "sysLocation" object in SNMPv2-MIB.

C H A P T E R
12 Interface Parameters
This chapter describes how to set the interfaces of the Aastra unit:
 How to reserve an IP address in a network server.
 Link Connectivity Detection
 Partial Reset
 Managing interfaces.
 PPPoE parameters.
 LLDP Configuration
 Ethernet Link Configuration
 DHCP Server Configuration
 Ethernet Connection Speed
 Configuring a MTU Value
Reserving an IP Address
Before connecting the Aastra unit to the network, Aastra strongly recommends that you reserve an IP address
in your network server – if you are using one – for the unit you are about to connect. This way, you know the
IP address associated with a particular unit.
Network servers generally allocate a range of IP addresses for use on a network and reserve IP addresses
for specific devices using a unique identifier for each device. The Aastra unit unique identifier is its media
access control (MAC) address. You can locate the MAC address as follows:
 It is printed on the label located on the bottom side of the unit.
 It is stored in the Device Info page of the web interface.
 You can take one of the telephones connected to the Aastra unit and dial *#*1 on the keypad.
The MAC address of the Aastra unit will be stated.
Aastra recommends to reserve an IP address with an infinite lease for each Aastra unit on the network.
Link Connectivity Detection
Each Ethernet port of the Aastra unit is associated with an Ethernet link. This information is available in the
Ethernet Ports Status section of the Network / Status page. A link has connectivity if at least one of its port
status is not disconnected.
The link connectivity is periodically polled (every 500 milliseconds). It takes two consecutive detections of the
same link state before reporting a link connectivity transition. This avoids reporting many link connectivity
transitions if the Ethernet cable is plugged and unplugged quickly.
Partial Reset
When a partial reset is triggered, the Rescue interface is configured and enabled with:
 its hidden IPv4 link configuration values
 its hidden IPv4 address configuration

Chapter 12 - Interface Parameters Interfaces Configuration
 an IPv6 link-local address on all network links

Hidden values are set by the unit's profile.
Just before the Rescue is configured, all IPv4 network interfaces that could possibly conflict with the Rescue
interface are disabled.
If the BNI Service is stopped when the partial reset occurs, it is started and the above configuration is applied.
Interfaces Configuration
Standards Supported • IEEE 802.1Q – Virtual Bridged Local Area Networks

• RFC 2460: IPv6
• RFC 2464: Transmission of IPv6 Packets over Ethernet
Networks
• RFC 4193: Unique_local_address
• RFC 4291: IP Version 6 Addressing Architecturea
• RFC 4443: ICMPv6
• RFC 4861: IPv6_neighbor_discovery
• RFC 4862: IPv6_stateless_autoconf
a. Site-local address are deprecated.
The Interface Configuration section allows you to add and remove up to 48 network interfaces. By default, this
section contains the following network interfaces:
 The Uplink interface, which defines the uplink information required by the Aastra unit to
properly connect to the WAN. The Uplink network interface is the IP interface that encapsulates
the following link interface (WAN connection):
• eth1 (TA7102i), wan for the Aastra TA7102i
By default, this interface uses the IPv4 DHCP connection type.
 The Rescue interface, which defines the address and network mask to use to contact the
Aastra unit after a partial reset operation. You cannot delete this interface. See “Partial Reset”
 The LAN interface IPv4 address and network mask.
The current status of the network interfaces is displayed in the Status page. It allows you to know which
interfaces are actually enabled. Enabled networks are activated when their configured link gets connectivity
and are deactivated as soon as the link connectivity is lost. See “Link Connectivity Detection” on page 85 for
more details.
The Interfaces Status section of the Status page displays the status of all currently enabled network interfaces,
including interfaces with an invalid configuration or waiting for a response.
When configuring network interfaces, Aastra recommends to have a syslog client properly configured and
enabled in order to receive any message related to the network interfaces behaviour. The interface used to
access the syslog client must also be properly enabled. See “Chapter 7 - Syslog Configuration” on page 71
for more details on enabling a syslog client.
Caution: Use extreme care when configuring network interfaces, especially when configuring the network
interface used to contact the unit for management. Be careful never to disable or delete the network interface
used to contact the unit. Also be careful to always set the unit’s management interface to be an interface
that you can contact.
Note: When performing a partial reset (see “Partial Reset” on page 15 for more details), the management
interface used for SNMP, CLI and WEB is automatically set to the Rescue interface. In that case, you must
change the Aastra unit system management network interface to something other than Rescue. Note that
you must be able to contact the interface you select.

Interfaces Configuration Software Configuration Guide
 To configure interfaces parameters:

1. In the web interface, click the Network link, then the Interfaces sub-link.
Figure 29: Network – Interfaces Web Page
3 4 5 6 7
2. If you want to add a new interface, enter its name in the blank field in the bottom left of the window,
then click the button.
The name is case-sensitive. Using the special value “All” is not allowed.
You can use the following ASCII codes in the network interface name:
49 1 77 M 103 g
50 2 78 N 104 h
51 3 79 O 105 i
52 4 80 P 106 j
53 5 81 Q 107 k
54 6 82 R 108 l
55 7 83 S 109 m
56 8 84 T 110 n
57 9 85 U 111 o
65 A 86 V 112 p
66 B 87 W 113 q
67 C 88 X 114 r
68 D 89 Y 115 s
69 E 90 Z 116 t
70 F 95 _, underscore 117 u
71 G 97 a 118 v
72 H 98 b 119 w
73 I 99 c 120 x
74 J 100 d 121 y
75 K 101 e 122 z
76 L 102 f
A valid network interface name must be compliant with the following rules:
• It must start with a letter
• It cannot contain characters other than letters, numbers, underscores.
If your Aastra unit contains an invalid interface name created in a previous firmware version without
the validation feature, the invalid interface name will be modified everywhere it appears on the first
restart and a syslog notification will be sent.
You can also delete an existing network interface by clicking the corresponding button. You
cannot delete the Rescue interface.
3. In the Interface Configuration section, select the link on which to activate the interface in the Link
column.
A VLAN is listed with the following syntax:
Link.VLAN ID
For instance, if you have added VLAN 20 on the interface eth5, it is listed as follows:
eth5.20

Chapter 12 - Interface Parameters Interfaces Configuration
Figure 30: VLAN Example
4. Select the configuration source of the interface information in the Type drop-down menu.
Table 46: Interface Configuration Sources
Source Description
IPv4 The IPv4 address and network mask are provided by querying a DHCP server and
DHCP using standard DHCP fields or options. Using the DHCP configuration assumes that
you have properly set your DHCP server with the relevant information. DHCP servers
may provide a list of IP configuration parameters to use. See “DHCP Server
Static every time the Aastra unit restarts. Use the static configuration if you are not using a
DHCP server/PPP peer or if you want to bypass it.
IPv4 IPv4 over PPP connection, address and network mask are provided by the PPP peer
PPPoE using IPCP. PPP peers may provide a list of IP configuration parameters to use. See
“PPPoE Configuration” on page 91 for more details.
IPv6 IPv6 state-less auto-configuration. See “IPv6 Autoconfiguration Interfaces” on page 89
Auto- for more details.
Conf
Static every time the Aastra unit restarts. Use the IPv6 static configuration if you are not using
IPv6 stateless or stateful auto-configuration or if you want to bypass it.
Note: If no network is configured in IPv6, the unit does not have any IPv6 address, not even the Link-Local
address. When a network is configured in IPv6, the Link-Local (FE80 ::...) address is automatically created
and displayed in the Network Status information.
5. If the interface configuration source is IPv4 Static or IPv6 Static, enter the address and network
mask (if applicable) of the network interface in the Static IP address field.
6. If the interface configuration source is IPv4 Static or IPv6 Static, set the Static Default Router field
with the IP address of the default gateway for the network interface.
7. Define whether or not the Aastra unit should attempt to activate the corresponding network interface
in the Activation drop-down menu.
page.
The current network interface information is displayed in the Status page.
Table 47: Network Interface Status
Status Description
Disabled The interface is not operational because it is explicitly disabled or the link
interface is unavailable.
Invalid Config The interface is not operational because its configuration is not valid.

Interfaces Configuration Software Configuration Guide
Table 47: Network Interface Status (Continued)
Status Description
Network Conflict The interface is configured with an IP address that is already used on the
network.
Link Down The interface is configured with a link that has no connectivity.
Waiting The interface is not operational because a response from a peer or server
Response is required.
Active The interface is operational.
IPv6 Autoconfiguration Interfaces

When the Type drop-down menu is set to IPv6 Auto-Conf, the network interface is an IPv6 over Ethernet
connection with IP parameters obtained by stateless auto-configuration or stateful (DHCPv6) configuration.
Autoconfiguration of IPv6 address is first initiated using state-less autoconfiguration. Stateful
autoconfiguration is initiated only if one of the following conditions is met:
 The router explicitly required stateful autoconfiguration by setting the “managed” or “other” flag
of the router advertisement.
 No router advertisement was received after 3 router solicitations. RFC 4861 defines the
number of router solicitations to send and the 4 seconds interval between the sent router
solicitations.
Stateless Autoconfiguration
All IPv6 addresses present in the router advertisements are applied to the network interface. Each IPv6
address is assigned a network name based on the configured network name with a suffix in the following
format: ConfiguredNetworkName-XX-Y.
XX is the address scope
 GU (Global Unique)
 UL (Unique Local)
 LL (Link-Local)
Y is a unique ID for the address scope.
Spanning Tree Protocol vs Stateless Autoconfiguration

Many network switches use the Spanning Tree Protocol (STP) to manage Ethernet ports activity. STP uses a
detection timeout before a router advertisement is sent to the Aastra unit. The default value for this timeout is
usually 30 seconds. However, when the unit wants to get an IPv6 address in Stateless autoconfiguration, this
timeout is too long and the unit falls into Stateful Autoconfiguration mode before it receives the router
advertisement. This results in the unit receiving a DHCPv6 address.
To solve the issue, check if the default STP detection timeout value in your router can be modified. If so, set
it to a value of 8 s or less. If you cannot modify the timeout value, Aastra recommends to disable the Spanning
Tree Protocol on the network to which the unit is connected.
Stateful Autoconfiguration
Stateful autoconfiguration is managed by DHCPv6. The DHCPv6 lease is negotiated according to RFC 3315
with the limitations listed in section 1.5. DHCPv6 may be used to obtain the following information (depending
on the router advertisement flags):
 IPv6 addresses (when the router advertisement “managed” flag is set)
 Other configuration (when the router advertisement “other” flag is set)
If only the “other” flag is set in the router advertisement, the DHCPv6 client only sends an information request
to the DHCPv6 server, otherwise it sends a DHCPv6 solicit message. If the flags change over time, only the
transitions from “not set” to “set“ are handled.

Chapter 12 - Interface Parameters Rescue Interface Configuration
Network Interface Priority

You can prioritize the network interfaces of the Aastra unit. In case of address conflicts between two or more
network interfaces, the network interface with the highest priority will remain enabled and the other interfaces
will be disabled. If the priority is the same, only the first enabled network interface will be able to use the IP
address. When a conflict ends, all network interfaces concerned automatically return to an operational state.
The actual status of network interfaces is displayed in the Status web page.
 To set the network interface priority:

1. In the ethMIB, set the networkInterfacesPriority variable with the proper value for the
corresponding interface.
eth.networkInterfacesPriority="Value"
where Value may be any number between 0 and 100.
Rescue Interface Configuration
You can define whether or not the Aastra unit should attempt to activate the rescue network interface.
Caution: Please be careful when using this section.
 To enable/disable the Rescue interface:

1. In the Rescue interface section, define whether or not the Aastra unit should attempt to activate the
corresponding network interface in the Activation drop-down menu.
Figure 31: Rescue Interface Configuration Section
page.

PPPoE Configuration Software Configuration Guide
PPPoE Configuration
Standards Supported • RFC 1332 – The PPP Internet Protocol Control Protocol
(IPCP)
• RFC 1334 – PPP Authentication Protocolsa
• RFC 1661 – The Point-to-Point Protocol (PPP)
• RFC 1877 – PPP Internet Protocol Control Protocol
Extensions for Name Server Addressesb
• RFC 1994 – Challenge Handshake Authentication Protocol
(CHAP)
• RFC 2516 – A Method for Transmitting PPP Over Ethernet
(PPPoE)
a. Section 2 (PAP), section 3 is obsoleted by RFC 1994
b. Supported except for sections 1.2 and 1.4
The PPPoE Configuration section applies only if you have selected the PPPoE connection type in the Interface
Configuration section of the web page.
 To configure PPPoE parameters:

1. In the PPPoE Configuration section, set the name of the service requested to the access
concentrator (AC) when establishing the next PPPoE connection in the Service Name field.
Figure 32: PPPoE Configuration Section
1
2
3
This is used as the Service-Name field of the packet broadcasted to the access concentrators. See
RFC 2516 section 5.1 for details.
The field may be set with any string of characters, with a maximum of 255 characters.
If you leave this field empty, the Aastra unit looks for any access concentrator.
2. Select the authentication protocol to use for authenticating the system to the PPP peer in the
Protocol drop-down menu.
• PAP: Use the Password Authentication Protocol.
• CHAP: Use the Challenge Handshake Authentication Protocol.
3. Set the PPP user name and password that identify the system to the PPP peer during the
authentication process in the User Name and Password fields.
Caution: The User Name and Password fields are not accessible if you have the User or Observer access
right. See “Users” on page 591 for more details.
When connecting to an access concentrator, it may request that the Aastra unit identifies itself with
a specific user name and password.
There are no restrictions, you can use any combination of characters.
The current PPPoE information is displayed in the Status page.
PPP Negotiation
When the Aastra unit restarts, it establishes the connection to the access concentrator in conformance with
the RFCs listed in “PPPoE Configuration” on page 91.

Chapter 12 - Interface Parameters PPPoE Configuration
When establishing a PPP connection, the Aastra unit goes through three distinct phases:
 Discovery phase
 Authentication phase
 Network-layer protocol phase
Discovery Phase
The Aastra unit broadcasts the value of the Service Name field.
The access concentrator with a matching service name answers the Aastra unit.
 If no access concentrator answers, this creates a “PPPoE failure” error.
 If more than one access concentrators respond to the discovery, the Aastra unit tries to
establish the PPP connection with the first one that supports the requested service name.
Authenthication Phase
If the access concentrator requests authentication, the Aastra unit sends the ID/secret pair configured in the
User Name and Password fields. If the access concentrator rejects the authentication, this creates an
“authentication failure” error.
Network-Layer Protocol Phase

The Aastra unit negotiates an IP address. The requested IP address is the one from the last successful PPPoE
connection. If the Aastra unit never connected by using PPPoE (or after a factory reset), it does not request
any specific IP address.
DHCP Client Identifier Presentation

You can define the method to use to present the value of the Client Identifier (Option 61) field through a DHCP
request. The following values are available:
Table 48: DHCP Client Identifier Presentation Parameters
Disabled The Client Identifier option is not presented in a DHCP request.

MacAscii The Client Identifier value is presented as the client MAC address in ASCII format. The MAC
address is represented in lowercase.
MacBinary The Client Identifier value is presented as the client MAC address in binary format.
 To define the DHCP client identifier presentation:

1. In the bniMIB, locate the DhcpClientGroup folder.
2. Set the dhcpClientIdentifierPresentation variable with the proper presentation.
bni.dhcpClientIdentifierPresentation="Value"

LLDP Configuration Software Configuration Guide

Table 49: DHCP Client Identifier Presentation Values
Value Meaning
100 Disabled
200 MacAscii
300 MacBinary
LLDP Configuration
The Link Layer Discovery Protocol (LLDP) service is used by network devices for advertising their identity,
capabilities, and neighbors on a IEEE 802 local area network, usually wired Ethernet.
The LLDP Configuration section allows you to configure parameters related to LLDP.
 To configure LLDP parameters:

1. In the LLDP Configuration section, set the network interface name on which LLDP should be
enabled in the Network Interface drop-down menu.
Figure 33: LLDP Configuration Section
1
2
3
LLDP cannot be activated on multiple network interfaces simultaneously.

2. Select the address type to populate the chassis ID device identifier in the Chassis ID drop-down
menu.
Table 50: Chassis ID Parameters
MAC Address The MAC address.

Network Address The IP address (or 0.0.0.0 if DHCP is not obtained yet).
3. Select whether to enable the LLDP-MED protocol override of the VLAN ID, User Priority and
DiffServ values in the Override Network Policy drop-down menu.
Table 51: Override Network Policy Parameters
Enable The service listens for LLDP advertisements, and overrides the previously
configured VLAN ID, User Priority and DiffServ with the values received.
Disable The service only publishes its characteristics and configurations by LLDP,
and does not override anything.
The LLDP-MED (Media Endpoint Discovery) protocol is an enhancement of LLDP.

The current LLDP information is displayed in the Status page.

Chapter 12 - Interface Parameters Ethernet Link Configuration
Ethernet Link Configuration
Standards Supported • IEEE 802.1X-2001 – Port Based Network Access Control
The Ethernet Link Configuration section allows you to configure the MTU as well as IEEE 802.1X
authentication.
 To configure Ethernet link parameters:

1. In the Ethernet Link Configuration section, set the MTU field of a specific Ethernet link with a proper
value.
Figure 34: Ethernet Link Configuration Section

1 2 3 4
The Maximum Transmission Unit (MTU) is a parameter that determines the largest packet than can
be transmitted by an IP interface (without it needing to be broken down into smaller units). Each
interface used by TCP/IP may have a different MTU value specified. See “Appendix C - Maximum
Transmission Unit (MTU)” on page 639 for more details on MTU.
The range is from 576 to 1500 bytes. All VLAN connections use the MTU size configured on their
related Ethernet link.
Note: The MTU value applied for a PPPoE connection is the smallest of the value negotiated with the server
and the value configured here.
2. Define the IEEE 802.1x authentication protocol activation to use for a specific Ethernet link in the
corresponding 802.1x Authentication drop-down menu.
802.1X Authentication is a tag optionally added to the Ethernet frame header to specify the support
of the IEEE 802.1X Authentication. It allows getting authorization and access to secured network(s).
Table 52: 802.1x Authentication Parameters
Disable The IEEE 802.1x authentication protocol is disabled on the Ethernet link interface.
Enable The IEEE 802.1x authentication protocol using the EAP-TLS authentication
method is enabled on the Ethernet link to get an access, through an IEEE 802.1x
EAP-TLS authenticator (such as an IEEE 802.1x capable network device), to
secured network(s). The Ethernet link interface remains always 'UP' whatever the
result of the IEEE 802.1x authentication.
3. Set the username used to authenticate each Ethernet link interfaces during the IEEE 802.1x EAP-
TLS authentication process in the corresponding EAP Username field.
This parameter is used only when the IEEE 802.1x authentication is enabled (802.1x Authentication
drop-down menu set to Enabled).
4. Define the IEEE 802.1x level of validation used by the device to authenticate the IEEE 802.1x EAP-
TLS peer's certificate.

DHCP Server Configuration Software Configuration Guide
This parameter also controls the criteria used to select the host certificate sent during the
authentication handshakes..
Table 53: 802.1x Certificate Validation Parameters
No No validation is performed on the peer’s certificate. Authentication with the peer is

Validation attempted even if the system time is not synchronized. If more than one host
certificate is configured for an EAP-TLS usage, the one with the latest expiration
date is used.
Trusted Allow a connection to the network by validating if the authentication peer’s
And Valid certificate is trusted and valid. The IEEE 802.1x authentication is attempted only if
the system time is synchronized. If more than one host certificate is configured for
an EAP-TLS usage, the one that is currently valid and with the latest expiration
date is used.

The current status of the network interfaces is displayed in the Status page. It allows you to know
which interfaces are actually enabled.
Table 54: Ethernet Link Interface State
State Description
Disconnected The link interface is physically disconnected.

Up The link interface is physically connected and considered as usable by network
interface(s).
DHCP Server Configuration
Standards Supported • RFC 2131 – Dynamic Host Configuration Protocola

• RFC 2132 – DHCP Options and BOOTP Vendor Extensionsb
• RFC 3315: DHCPv6c
a. Supports the client side of the protocol
b. Only sections 3.3, 3.5, 3.8 and 8.3
c. Supports the client side of the protocol
Note: This section applies only if you are using the DHCP connection type (“Interfaces Configuration” on
page 86).
DHCP servers generally allocate a range of IP addresses for use on a network and reserve IP addresses for
specific devices using a unique identifier for each device. The Aastra unit unique identifier is its media access
control (MAC) address.
You can locate the MAC address as follows:
 on the label located on the bottom side of the unit.
 in the System > Information web page
 you can dial the following digits on a telephone connected to the Aastra unit:
*#*1
The Aastra unit answers back with its MAC address. This applies to units with FXS interfaces. See
“General POTS Configuration” on page 160 for more details.
Aastra recommends to reserve an IP address with an infinite lease for each Aastra unit on the network.

Chapter 12 - Interface Parameters DHCP Server Configuration
DHCP Negotiation
The DHCP lease is negotiated according to RFC 2131 (supports the client side of the protocol) and RFC 2132
(only sections 3.3, 3.5, 3.8 and 8.3). The following paramaters are set
Table 55: DHCP Parameters
DHCP Parameter Value
Host Name (option 12) Set according to the Host Name parameter of the Network > Host
page (“Host Configuration” on page 89). This option cannot be empty
according to RFC 2132. If the Host Name parameter is empty, the
DHCP option 12 is not sent.
Vendor Class Identifier (option 60) Set according to the System Description parameter of the System >
Information page.
Client identifier (option 61) Set according to MAC Address parameter of the System >
Information.

Ethernet Connection Speed Software Configuration Guide
Ethernet Connection Speed
You can set the speed and duplex of the Ethernet connection of the Aastra unit. The following values are
available:
Table 56: Ethernet Ports Speed and Duplex Supported
Auto Automatic negociation of speed and duplex.

Half10 10 Mbit/s Half-duplex.
Full10 10 Mbit/s Full-duplex.
Half100 100 Mbit/s Half-duplex.
Full100 100 Mbit/s Full-duplex.
A half-duplex connection refers to a transmission using two separate channels for transmission and reception,
while a full-duplex connection refers to a transmission using the same channel for both transmission and
reception.
If unknown, set the variable to Auto so that the Aastra unit can automatically detect the network speed.
Caution: Whenever you force a connection speed / duplex mode, be sure that the other device and all
other intermediary nodes used in the communication between the two devices have the same configuration.
See “Speed and Duplex Detection Issues” on page 98 for more details.
The current speed and duplex configuration is displayed in the Network > Status page under the Ethernet
Ports Status section.
 To set the Ethernet connection speed and duplex:

1. In the ethMIB, locate the portsTable folder.
2. Set the portsSpeed variable with the proper Ethernet speed and duplex.
eth.portsSpeed="Value"

Table 57: Ethernet Ports Speed and Duplex Values
Value Meaning
100 Auto
200 Half10
300 Full10
400 Half100
500 Full100

Chapter 12 - Interface Parameters Ethernet Connection Speed
Speed and Duplex Detection Issues

There are two protocols for detecting the Ethernet link speed:
 An older protocol called parallel detection.
 A more recent protocol called auto-negotiation (IEEE 802.3u).
The auto-negotiation protocol allows to detect the connection speed and duplex mode. It exchanges
capabilities and establishes the most efficient connection. When both endpoints support the auto-negotiation,
there are no problems. However, when only one endpoint supports auto-negotiation, the parallel detection
protocol is used. This protocol can only detect the connection speed; the duplex mode cannot be detected. In
this case, the connection may not be established.
The Aastra unit has the possibility to force the desired Ethernet link speed and duplex mode by disabling the
auto-negotiation and selecting the proper setting. When forcing a link speed at one end, be sure that the other
end (a hub, switch, etc.) has the same configuration. To avoid any problem, the link speed and duplex mode
of the other endpoint must be exactly the same.

C H A P T E R
13 VLAN Parameters
This chapter describes how to create and manage dynamic VLANs on the Aastra unit.
VLAN Configuration
A virtual LAN is a network of computers that behave as if they are connected to the same wire even though
they may actually be physically located on different segments of a LAN. You can add VLANs on the Ethernet
links of the Aastra unit. You can currently add or manage up to a maximum of 16 VLANs.
Caution: When working with VLANs, take care not to cut your access to the unit, for instance by putting
the Uplink on a VLAN to which your PC does not have access and then setting the management interface
to Uplink.
 To add a VLAN:
1. In the web interface, click the Network link, then the VLAN sub-link.
Figure 35: Network – VLAN Web Page
2 3 5 4
2. Select the Ethernet link over which the VLAN interface is built in the Link drop-down menu.
3. Set the VLAN ID used by the VLAN interface in the Id field.
This is a 12 bit field in the 802.1Q tag carrying an ID that differentiates frames containing this ID
from frames containing different IDs or no 802.1Q tag at all.
To systems supporting Ethernet 802.1Q, frames containing the same VLAN ID are considered as
belonging to the same virtual LAN, and frames containing different IDs are considered as not
belonging to the same virtual LAN, even though they use the same physical LAN.
4. Click on the button.
5. Set the default user priority value the interface uses when tagging packets in the Default User
Priority field.
You can also set specific service class values in the Quality of Service page. See “Chapter 14 -
Local QoS (Quality of Service) Configuration” on page 115 for more details.
You can also delete an existing VLAN by clicking the corresponding button.
Once you have added a VLAN, you must select this VLAN on an interface to activate it. You can do
so in the Link column of the Interface Configuration section in the Network > Interfaces page
(“Interfaces Configuration” on page 100). The VLAN is listed with the following syntax:
Link.VLAN ID
For instance, if you have added VLAN 20 on the interface eth5, it is listed as follows:
eth5.20

Chapter 13 - VLAN Parameters VLAN Configuration
Figure 36: VLAN Example

C H A P T E R
14 Local QoS (Quality of Service)

Configuration
This chapter describes how to configure the local QoS parameters. The local QoS tags packets sent from the
Aastra unit. It does not process nor classify packets coming from the network.
Introduction
QoS (Quality of Service) features enable network managers to decide on packet priority queuing. The Dgw
v2.0 application supports the Differentiated Services (DS) field and 802.1q taggings.
The Dgw v2.0 application supports the Real Time Control Protocol (RTCP), which is used to send packets to
convey feedback on quality of data delivery.
The Dgw v2.0 application does not currently support the Voice Band Data service class. It also does not
support RSVP (Resource Reservation Protocol).
Differentiated Services (DS) Field
Standards Supported RFC 2475: An Architecture for Differentiated Services
Differentiated Services (DiffServ, or DS) is a protocol for specifying and controlling network traffic by class so
that certain types of traffic – for example, voice traffic, which requires a relatively uninterrupted flow of data,
might get precedence over other kinds of traffic.
What are Differentiated Services?

Differentiated Services avoids simple priority tagging and depends on more complex policy or rule
statements to determine how to forward a given network packet. An analogy is made to travel services, in
which a person can choose among different modes of travel – train, bus, airplane – degree of comfort,
the number of stops on the route, standby status, the time of day or period of year for the trip, and so
forth.
For a given set of packet travel rules, a packet is given one of 64 possible forwarding behaviors – known
as per hop behaviors (PHBs). A six-bit field, known as the Differentiated Services Code Point (DSCP), in
the Internet Protocol header specifies the per hop behavior for a given flow of packets. The DS field
structure is presented below:
0 1 2 3 4 5 6 7
+---+---+---+---+---+---+---+---+
| DSCP | CU |
+---+---+---+---+---+---+---+---+
MSB LSB
• DSCP: Differentiated Services CodePoint.

• CU: Currently Unused. The CU bits should always be set to 0.
For both signalling and media packets, the DSCP field is configurable independently. The entire DS field
(TOS byte) is currently configurable.

Chapter 14 - Local QoS (Quality of Service) Configuration Differentiated Services (DS) Field
DiffServ replaces the first bits in the ToS byte with a differentiated services code point (DSCP). It uses the
existing IPv4 Type of Service octet.
It is the network administrator’s responsibility to provision the Aastra unit with standard and correct values.
 To configure the Aastra unit DiffServ value:

1. In the web interface, click the Network link, then the QoS sub-link.
Figure 37: Network – QoS Web Page
2
3
2. Set the default Differentiated Services value used by the unit for all generated packets in the Default
DiffServ (IPv4) field.
You can override this value by setting specific service class values. See “Specific Service Class
This 8-bit value is directly set in the TOS field (2nd byte) of the header of transmitted IPv4 packets,
allowing you to use either DiffServ or TOS mapping.
The DiffServ value is 1 octet scalar ranging from 0 to 255. The DSCP default value should be
101110. This results in the DS field value of 10111000 (184d). This default value would result in a
value of “101” precedence bits, low delay, high throughput, and normal reliability in the legacy IP
networks (RFC 791, RFC 1812). Network managers of legacy IP networks could use the above-
mentioned values to define filters on their routers to take advantage of priority queuing. The default
value is based on the Expedited Forwarding PHB (RFC 2598) recommendation.
Note: RFC 3168 now defines the state in which to set the two least significant bits in the TOS byte. On the
other hand, this RFC only applies to TCP transmissions and the bits are thus set to “0” in the Aastra unit.
This has the following effects:
• The TOS values for UDP packets are the same as in the MIB.
• The TOS values for TCP packets are equal to the closest multiple of 4 value that is not greater than the
value in the MIB.
You can find references on DS field under the IETF working group DiffServ. For more information,
please refer to the following RFC documents:
• Definition of the Differentiated Services Field (DS Field) in the IPv4 and IPv6 Headers
(RFC 2474)
• An Architecture for Differentiated Services (RFC 2475)
• Assured Forwarding PHB Group (RFC 2597)
• An Expedited Forwarding PHB (RFC 2598)
3. Set the Default Traffic Class value used by the unit for all generated IPv6 packets in the Default
Traffic Class (IPv6) field.
Specific service class values may be set in the Service Classes table. See “Specific Service Class
The 8-bit Traffic Class field in the IPv6 header is available for use by originating nodes and/or
forwarding routers to identify and distinguish between different classes or priorities of IPv6 packets.

IEEE 802.1q Software Configuration Guide
IEEE 802.1q
The 802.1q standard recommends the use of the 802.1q VLAN tags for Ethernet frames traffic prioritization.
VLAN tags are 4-byte headers in which three bits are reserved for priority indication. The values of the priority
bits shall be provisioned.
The 802.1q standard comprises the 802.1p standard.
It is the network administrator’s responsibility to provision the Aastra unit with standard and correct values.
 To enable the IEEE 802.1q user priority configuration:

1. In the Ethernet 802.1Q Tagging Configuration section of the QoS page, select Enable in the Enable
column for each interface on which you want to enable user priority tagging.
Figure 38: Ethernet 802.1Q Tagging Configuration Section

1 2
The VLAN ID part of the 802.1Q tag is always set to 0.

2. Set the default user priority value each interface uses when tagging packets in the Default User
Priority column.
You can override each value by setting specific service class values. See “Specific Service Class
The user priority is a 3 bit field in the 802.1Q tag that carries a priority value ranging from 0 to 7 and
may be used by switches to prioritize traffic. The 802.1q default priority value should be 6 for both
signalling and media packets.
Specific Service Class Configuration
You can override the default value set in the DiffServ and 802.1q sections for each service class of the Aastra
unit:
 Signalling
 Voice
 T.38
 IP Sync (IP Sync is not available in IPv6)
 To set specific service class values:

1. In the Service Class Configuration section of the QoS page, set a specific DiffServ value for each
class in the DiffServ (IPv4) column.
Figure 39: Service Class Configuration Section

1 2 3
See “Differentiated Services (DS) Field” on page 101 for more details.
2. Set the Default Traffic Class value used in IPv6 packets for each class in the Traffic Class (IPv6)
column.

Chapter 14 - Local QoS (Quality of Service) Configuration Network Traffic Control Configuration
The 8-bit Traffic Class field in the IPv6 header is available for use by originating nodes and/or
forwarding routers to identify and distinguish between different classes or priorities of IPv6 packets.
3. Set a specific user priority for each class in the User Priority column.
See “IEEE 802.1q” on page 103 for more details.
Network Traffic Control Configuration
You can apply a bandwidth limitation on the network interfaces. The limitations are applied on raw data on the
physical link and not only on the payload of the packets. All headers, checksums and control bits (TCP, IP,
CRC, etc.) are considered in the actual bandwidth.
A bandwidth limitation is applied on a physical link and not on a high-level network interfaces. All high-level
network interfaces (including VLANs) using the same physical link are affected by a configured limitation. This
limitation is applied egress only (outgoing traffic).
If the NTC service is stopped, this section is not displayed in the QoS page. See “Chapter 4 - Services” on
page 53 on information on how to start the service. Starting the NTC service enables Traffic Shaping even if
bandwidth limitation is disabled.
Bandwidth limitation is an average of the amount of data sent per second. It is thus normal that the unit sends
a small burst of data after a period of silence.
Note that the NTC service sends packets on the physical link according to their respective priorities as
described below. Lower priority packets are dropped first.
Table 58: Physical Link Priorities
Priority Description
1 Highest priority. Packets originating from the unit with 802.1p priority set to 7.
2 Packets originating from the unit with 802.1p priority set to 6.
9 Lowest priority. Packets originating from another link interface (routed packets).
Packets that exceed the defined bandwidth are eventually dropped (when the buffers are exceeded). This
implies that data bursts can suffer a slight amount of packet loss. The different codecs configured and the
desired number of simultaneous channels should be taken into account when choosing a bandwidth limit to
prevent call drops, choppy voice or inconstant ptime. The NTC service can impact the execution of other
processes if the number of packets to process is too high. (High traffic and/or low limit).
 To set network traffic control parameters:

1. In the Network Traffic Control Configuration section of the QoS page, set the corresponding Egress
Limit field with the egress bandwidth limitation for the selected link interface.
The range is from 64 to 40960 kilobits per second.
The value 0 means no bandwidth limitation and no prioritization.

Network Traffic Control Configuration Software Configuration Guide
This value must be set according to the upstream bandwidth limit of the network on this link. Set to
0 (disable) if the network bandwidth exceeds 40960 kbps or if it exceeds the effective limit of this
device.
Figure 40: Network Traffic Control Configuration Section

1

Chapter 14 - Local QoS (Quality of Service) Configuration Network Traffic Control Configuration

C H A P T E R
15 Local Firewall Configuration
This chapter describes how to configure the local firewall parameters.

 Setting the default policy
 Creating/editing a firewall rule
 Moving a firewall rule
 Deleting a firewall rule
 Disabling the local firewall
Managing the Local Firewall
The local firewall allows you to dynamically create and configure rules to filter packets. The traffic is analyzed
and filtered by all the rules configured.
Note: The Aastra unit’s local firewall settings do not support IPv6. See “IPv4 vs. IPv6” on page 85 for more
details.
Since this is a local firewall, rules apply only to incoming packets with the unit as destination.
Incoming packets for an IP communication established by the unit are always accepted (Example : If the
Aastra unit sends a DNS request, the answer will be accepted).
Rules priority is determined by their position in the table.
The maximum number of rules allowed in the configuration is 20.
Caution: Enabling the local firewall and adding rules has an impact on the Aastra unit’s overall
performance as the firewall requires additional processing. The more rules are enabled, the more overall
performance is affected. Furthermore, Aastra recommends to use a 30 ms packetization time when the
firewall is enabled (instead of a 20 ms ptime, for instance) in order to simultaneously use all the channels
available on the unit.
Partial Reset
When a partial reset is triggered and the firewall is enabled, the configuration is rolled back if it was being
modified. A new rule is then automatically applied in the firewall to allow access to the 'Rescue' interface.
However, if the firewall is disabled, the configuration is rolled back but no rule is added.
Setting the Default Policy

The default policy defines the action the Aastra unit must take when a packet does not match any rule.

Chapter 15 - Local Firewall Configuration Managing the Local Firewall
 To set the default policy:

1. In the web interface, click the Network link, then the Local Firewall sub-link.
Figure 41: Network – Local Firewall Web Page
2. In the Local Firewall Configuration section, define the Default Policy drop-down menu.
Table 59: Default Policy Parameters
Accept Lets the packet through.

Drop Drops the packet without any notification.
Caution: Make sure there are some rules with the Action parameter set to Accept in the local firewall
BEFORE applying changes that set the default policy to Drop. If you do not comply with this warning, you
will lose contact with the unit and a partial or factory reset will be required.
Setting the default policy to Drop or adding a rule automatically enables the local firewall. Enabling
the local firewall may have a negative impact on performance.
Creating/Editing a Firewall Rule

The web interface allows you to create a firewall rule or modify the parameters of an existing one.
 To create or edit a firewall rule:

1. In the Local Firewall Rules section of the Local Firewall page, do one of the following:
• If you want to add a rule before an existing entry, locate the proper row in the table and
click the button of this row.
• If you want to add a rule at the end of the existing rows, click the button at the
bottom right of the section.
Figure 42: Local Firewall Rules Section

2 3 4 5 6 7 8
Note: When you add a new rule, edit an existing rule, or delete a rule, you can see a yellow Yes in the
Config Modified section at the top of the window. It warns you that the configuration has been modified but
not applied (i.e., the Firewall section of the Status page differs from the Local Firewall). The Local Firewall
sub-menu is a working area where you build up a local firewall configuration. While you work in this area,
the configured parameters are saved but not applied (i.e., they are not used to filter incoming packets). The
yellow Yes flag warns you that the configuration has been modified but is not applied.

Managing the Local Firewall Software Configuration Guide
2. Set the current activation state for this rule in the corresponding Activation drop-down menu.
Table 60: Firewall Rule Activation State Parameters
Enable This rule is active in the firewall.

Disable This rule is not in the firewall.
Only enabled rules may be applied to the firewall.

3. Enter the source address of the incoming packet in the corresponding Source Address field.
Use one of the following syntax:
Table 61: Source Address Parameters
address[/mask] Can either be a network IP address (using /mask) or one of the host
IP addresses. The mask must be a plain number specifying the
number of binary 1s at the left side of the network mask (a mask of
24 specifies a network mask of 255.255.255.0).
networkInterfaceName The value must already exist in the Interface Configuration table
/ (see “Interfaces Configuration” on page 100 for more details). The
interface name is case sensitive, hence it must be entered properly.
If the specified network interface is disabled or removed, the rule is
automatically disabled thus removed from the firewall. When the
network interface is enabled or added back, the rule is automatically
enabled and applied in the firewall.
Note: It is mandatory to use the suffix “/” to indicate that the network
address of this interface is used instead of the host address.
Leaving the default empty string matches any address.

4. Enter the source port of the incoming packet in the corresponding Source Port field.
You can enter a single port or a range of ports. In the case of a range of ports, use the following
format:
port[-port]
Leaving the default empty string means that no filtering is applied on the source port, thus matching
any port.
This parameter is only effective when the Protocol drop-down menu is set to TCP or UDP (see Step
7).
5. Enter the destination address of the incoming packet in the corresponding Destination Address
field.
Table 62: Source Address Parameters
address Must be one of the host IP addresses. Specifying a network address

is invalid since this is a local firewall.

Chapter 15 - Local Firewall Configuration Managing the Local Firewall
Table 62: Source Address Parameters (Continued)
networkInterfaceName The host address of this interface is used. The value must already
exist in the Interface Configuration table (see “Interfaces
Configuration” on page 100 for more details). The interface name is
case sensitive, hence it must be entered properly.
If the specified network interface is disabled or removed, the rule is
automatically disabled thus removed from the firewall. When the
network interface is enabled or added back, the rule is automatically
enabled and applied in the firewall.

6. Enter the destination port of the incoming packet in the corresponding Destination Port field.
You can enter a single port or a range of ports. In the case of a range of ports, use the following
format:
port[-port]
Leaving the default empty string means that no filtering is applied on the destination port, thus
matching any port.
7).
7. Select the protocol of the incoming packet to filter in the corresponding Protocol drop-down menu.
Table 63: Firewal Rule Protocol Parameters
All Matches packets using any protocols.

TCP Matches only TCP packets.
UDP Matches only UDP packets.
ICMP Matches only ICMP packets.
8. Select the action to take in the corresponding Action field.

Table 64: Firewal Rule Action Parameters

Reject Sends back an ICMP port unreachable in response to the matched packet. The
packet is then dropped.
Note that if a connection is already established before creating a rule that rejects it, this connection
stays active despite the rule applied.
9. Click the Apply button to activate the enabled rules.
The current enabled rules applied are displayed in the Network > Status web page, Firewall section,
which contains the active configuration in the firewall. You can also see that the yellow Config
Modified Yes flag is cleared.
Moving a Firewall Rule

The firewall rules sequence is very important because rules priority is determined by their position in the table.
If you want the unit to try to match one rule before another one, you must put that rule first.

Disabling the Local Firewall Software Configuration Guide
 To move a rule up or down:

1. Either click the or arrow of the rule you want to move until the entry is properly located.
2. Click the Apply button to update the Network > Status web page.
Deleting a Firewall Rule

You can delete a rule from the table in the web interface.

1. Click the button of the rule you want to move.
Disabling the Local Firewall
When the local firewall is enabled, it has an impact on the Aastra unit’s overall performance as the firewall
requires CPU power. You can disable the firewall if you do not need it, thus not impacting performance.
 To disable the firewall:

1. In the Local Firewall Configuration section, set the default policy to Accept with no rules in the local
firewall.
2. Restart the Aastra unit.

Chapter 15 - Local Firewall Configuration Disabling the Local Firewall

C H A P T E R
16 IP Routing Configuration
This chapter describes how to configure the IP Routing parameters of the Aastra unit.
 IPv4 Forwarding
 Creating/editing an IP routing rule
 Moving an IP routing rule
 Deleting an IP routing rule
 IP routing examples
Managing IP Routing
The IP Routing service allows the Aastra unit to perform advanced routing based on the packet’s criteria
(source IP address and source Ethernet link), which allows the packet to be forwarded to a specific network.
You can create up to four advanced IP routes.
Note: The Aastra unit’s IP Routing settings do not support IPv6. See “IPv4 vs. IPv6 Availability” on page 85
for more details.
Packets matching a list of criteria should1 use advanced IP routes instead of routes present in the main routing
table of the unit.
IP Routing works together with the following services:
 Network Firewall (“Chapter 17 - Network Firewall Configuration” on page 135)
 NAT (“Chapter 18 - NAT Configuration” on page 141)
 DHCP server (“Chapter 19 - DHCP Server Settings” on page 149)
 Network Traffic Control (“Network Traffic Control Configuration” on page 118)
These services must be properly configured.
When the IP Routing service is started, IP routing is activated even if there is no configured rule (the Aastra
unit will forward received packets). If the IP Routing service is stopped, IP forwarding is disabled, this tab is
greyed out and the parameters are not displayed. See “Chapter 4 - Services” on page 53 on information on
how to start the service.
Caution: Enabling the IP routing service and adding rules has an impact on the Aastra unit’s overall
performance as IP routing requires additional processing. The more rules are enabled, the more overall
performance is affected. Furthermore, Aastra recommends to use a 30 ms packetization time when IP
routing is enabled (instead of a 20 ms ptime, for instance) in order to simultaneously use all the channels
IPv4 Forwarding
IPv4 forwarding allows you to control the IPv4 forwarding feature and the Advanced IP Routes. When set to
Enabled, IPv4 Forwarding is enabled and the Advanced IP Routes are applied. When set to Disabled, IPv4
Forwarding is disabled and the Advanced IP Routes are not applied (the Advanced IP Routes section of the
IP Routing page is disabled).
1. A packet matching a route uses the custom routing table first and then the main routing table if no route in the custom routing table was
able to send the packet to the desired destination IP address.

Chapter 16 - IP Routing Configuration Managing IP Routing
 To manage IPv4 forwarding:

1. In the web interface, click the Network link, then the IP Routing sub-link.
2. In the IP Routing Configuration section of the IP Routing page, define whether or not IPv4
forwarding is enabled by setting the IPv4 Forwarding drop-down menu accordingly.
Figure 43: IPv4 Forwarding Configuration Section
3. Click the Submit & Apply button to update the Network > Status web page.
Creating/Editing an IP Routing Rule

The web interface allows you to create a routing rule or modify the parameters of an existing one.
 To create or edit a routing rule:

1. In the Advanced IP Routes section of the IP Routing page, do one of the following:
Note: When you add a new rule, edit an existing rule or delete a rule, you can see a yellow Yes in the Config
Modified section at the top of the window. It warns you that the configuration has been modified but not
applied (i.e., the Advanced IP Routes section of the Status page differs from the IP Routing page). The IP
Routing sub-menu is a working area where you build up a routing configuration. While you work in this area,
the configured parameters are saved but not applied (i.e., they are not used to route packets). The yellow
Yes flag warns you that the configuration has been modified but is not applied.
Figure 44: Advanced IP Routes Section

2 3 4 5
2. Set the required state for this rule in the corresponding Activation drop-down menu.
Table 65: IP Routing Rule Activation Parameters
Enable Activates this route.

Disable Does not activate this route.
Only enabled rules may be applied to the routing table.

3. Enter the source IP address criteria an incoming packet must have to match this rule in the Source
Address field.

Managing IP Routing Software Configuration Guide
Use the following syntax:

Table 66: Source Address Syntax
Syntax Description
address[/mask] Can either be a network IP address (using /mask) or one of the

host IP addresses. The mask must be a plain number specifying
the number of binary 1s at the left side of the network mask (a
mask of 24 specifies a network mask of 255.255.255.0). For
instance:
• 192.168.0.11
• 192.168.1.0/24
networkInterfaceName[/] The value must already exist in the Interface Configuration table
(see “Interfaces Configuration” on page 100 for more details).
The interface name is case sensitive, hence it must be entered
properly.
If the specified network interface is disabled or removed, the rule
is automatically disabled thus removed from the NAT. When the
network interface is enabled or added back, the rule is
automatically enabled and applied. For instance:
• Lan1/ (Lan1 network address)
Note: It is mandatory to use the suffix “/” to indicate that the
network address of this interface is used instead of the host
address.
When left empty, any source address matches this rule.

4. Enter the source link criteria an incoming packet must have to match this rule in the Source Link
field.
When left empty, packets received on any link match this rule.
5. Select the network on which the packet is forwarded in the Forward to Network drop-down menu.
6. Click the Submit & Apply button to activate the enabled rules.
The current applied rules applied are displayed in the Network > Status web page, Advanced IP
Routes section, which contains the active configuration of the custom routing tables. You can also
see that the yellow Config Modified Yes flag is cleared.
Note: You can revert back to the configuration displayed in the Status web page at any time (including the
disabled rules) by clicking the Rollback button at the bottom of the page. All modified settings in the IP
Routing page will be lost.
Moving an IP Routing Rule

The IP routing rules sequence is very important because only one forwarding rule is applied on a packet. Rules
priority is determined by their position in the table. If you want the unit to try to match one rule before another
one, you must put that rule first.

Deleting an IP Routing Rule


Chapter 16 - IP Routing Configuration Managing IP Routing

Static IPv4 Routes

You can add or delete static IPv4 routes in the Aastra unit. A "static" route means that the route is configured
manually by the administrator. It can be configured through two different methods: through unit provisioning
or through a DHCP server (“DHCPv4 Classless Static Route Option” on page 117).
 To manage static IPv4 routes:

1. In the Static IP Routes section of the IP Routing page, do one of the following:
• If you want to add a route, click the button at the bottom of the section.
• If you want to delete an existing route, click the button of the route you want to
move.
Figure 45: Static IP Routes Section

2 3 4
This section is not available if IPv4 forwarding is disabled.

2. Specify the destination IP address criteria that an outgoing packet must have to match this route in
the corresponding Destination field.
The supported format for the destination is:
IP address[/mask]
When specifying a network as a destination, it is mandatory to use the "/" format.
The mask must be a plain number specifying the number of binary 1s at the left side of the network
mask (a mask of 24 specifies a network mask of 255.255.255.0). For instance:
• 192.168.1.5 specifies an IP address as the destination.
• 192.168.1.0/24 specifies a network address as the destination.
3. Select the output link (interface) name in the corresponding Link drop-down menu.
When left empty, the link is selected automatically according to the information already present in
the routing table.
4. Define the IP address of the gateway used by the route in the corresponding Gateway field.
The current routes available are displayed in the Network > Status web page, IPv4 Routes section.
This section identifies the entity that installed the route.
Table 67: IPv4 Routes Protocol
Protocol Description
Dhcp The route was installed dynamically by the DHCP protocol.

Static The route was installed by the administrator of the unit.
Kernel The route was installed by the operating system.
Other The route was installed by another entity.

Network Configuration Examples Software Configuration Guide
DHCPv4 Classless Static Route Option

Standards Supported • RFC 3442: The Classless Static Route Option for Dynamic
Host Configuration Protocol (DHCP) version 4
You can define whether or not the Classless Static Route Option is enabled. Static routes can be configured
through the Classless Static Route Option for DHCPv4 (option 121) defined in RFC 3442.
If a static route to 0.0.0.0/0 is received through option 121 while a default router is also specified (see “Default
Gateway Configuration” on page 91 for more details), the route received through option 121 has priority.
Table 68: DHCPv4 Classless Static Route Option Parameters
Request The device requests the Classless Static Route Option 121.
None Routes received from the DHCP server are ignored.
 To define define whether or not the Classless Static Route Option is enabled:
1. In the bniMIB, locate the DhcpClientGroup folder.
2. Set the dhcpClientClasslessStaticRouteOption variable with the proper behaviour.
bni.dhcpClientClasslessStaticRouteOption="Value"

Table 69: DHCPv4 Classless Static Route Option Values
Value Meaning
100 None
200 Request
Network Configuration Examples
The following are two examples of advanced IP routing that can be accomplished with the Aastra unit.
Forward Packets from the Lan1 Network to the Uplink Network with NAT
1. Create an IP routing rule so that the packets are routed (“Managing IP Routing” on page 113).
• Source IP: Lan1/
Remove this criterion if you want to forward all packets received on the lan link.
• Source Link: lan2
• Destination Network: Uplink
• Click Submit & Apply.
2. The source link name may vary depending on the unit model you have.

Chapter 16 - IP Routing Configuration Network Configuration Examples
2. Create a NAT rule so that the forwarded packets going on the Uplink network use the correct source
IP address (“Creating/Editing a Source NAT Rule” on page 141).
• Type: SNAT
• Protocol: All
• New Address: Uplink
3. Create a Network Firewall rule to let established or related packets go through the unit (if the default
policy is not set to Accept) (“Managing the Network Firewall” on page 135).
• Connection State: Established or Related
• Action: Accept
4. Create a Network Firewall rule to let the packets pass from the Lan1 network to the Uplink network
(if the default policy is not set to Accept). All response packets will be accepted by the previous rule
(“Managing the Network Firewall” on page 135).
Use additional rules or set the default policy to Accept if you want to forward packets
received on the lan link with a source address that does not match the Lan1 subnet.
• Connection State: New
• Action: Accept
Configure Port Forwarding for a Web Server Located on the LAN

1. Make sure the IP Routing service is started (to activate IP forwarding).
2. Create a NAT rule (“Creating/Editing a Destination NAT Rule” on page 145).
This will change the destination of an HTTP packet originally destined to the Aastra unit with the
IP:Port of the Web server on the LAN side (to make sure the unit does not process the packet but
forwards it on the Lan1 network).
• Type: DNat
• Destination IP: Uplink
• Destination Port: 8080
• Protocol: TCP
• New Address: 192.168.0.11:80 (IP:Port of the Web server on the LAN side)
3. Create a NAT rule (“Creating/Editing a Source NAT Rule” on page 141).
This will change the source IP address of the packet before it is sent on the Lan1 network (to make
sure the Web browser can reply correctly to the request).
• Type: SNat
• Destination IP: 192.168.0.11
• Protocol: TCP
• New Address: Lan1
4. Create a Network Firewall rule to let established or related packets go through the unit (if the default
policy is not set to Accept) (“Managing the Network Firewall” on page 135).
• Connection State: Established or Related
• Action: Accept

Network Configuration Examples Software Configuration Guide
5. Create a Network Firewall rule to let the packets pass from the Uplink network to the Lan1 network
(if the default policy is not set to Accept). All response packets will be allowed by the previous rule
(“Managing the Network Firewall” on page 135).
• Destination IP: 192.168.0.11
• Protocol: TCP
• Action: Accept

Chapter 16 - IP Routing Configuration Network Configuration Examples

C H A P T E R
17 Network Firewall Configuration
This chapter describes how to configure the network firewall parameters.

 Setting the default policy
 Creating/editing a firewall rule
 Moving a firewall rule
 Deleting a firewall rule
 Disabling the network firewall
Managing the Network Firewall
The network firewall allows dynamically creating and configuring rules to filter packets forwarded by the unit.
Since this is a network firewall, rules only apply to packets forwarded by the unit. The traffic is analyzed and
filtered by all the rules configured.
Note: The Aastra unit’s network firewall settings do not support IPv6. See “IPv4 vs. IPv6 Availability” on
If no rule matches the incoming packet, the default policy is applied. A rule's priority is determined by its index
in the table.
Rules using Network Names are automatically updated as the associated IP addresses and network mask are
modified.
If the Network Firewall service is stopped, all forwarded traffic is accepted, this tab is greyed out and the
parameters are not displayed. See “Chapter 4 - Services” on page 53 on information on how to start the
service.
The maximum number of rules allowed in the configuration is 20.
Caution: Enabling the network firewall and adding rules has an impact on the Aastra unit’s overall
performance as the firewall requires additional processing. The more rules are enabled, the more overall
performance is affected. Furthermore, Aastra recommends to use a 30 ms packetization time when the
firewall is enabled (instead of a 20 ms ptime, for instance) in order to simultaneously use all the channels
Setting the Default Policy

The default policy defines the action the Aastra unit must take when a forwarded packet does not match any
rules.

Chapter 17 - Network Firewall Configuration Managing the Network Firewall
 To set the default policy:

1. In the web interface, click the Network link, then the Network Firewall sub-link.
Figure 46: Network – Network Firewall Web Page
2. In the Network Firewall Configuration section, define the default policy in the Default Policy drop-
down menu.
Table 70: Default Policy Parameters

Setting the default policy to Drop or adding a rule automatically enables the network firewall.
Enabling the network firewall may have a negative impact on performance.
Creating/Editing a Network Firewall Rule

The web interface allows you to create a network firewall rule or modify the parameters of an existing one.
 To create or edit a network firewall rule:

1. In the Network Firewall Rules section of the Network Firewall page, do one of the following:
Figure 47: Network Firewall Rules Section

2 3 4 5 6 7 8 9
applied (i.e., the Firewall section of the Status page differs from the Network Firewall page). The Network
Firewall page is a working area where you build up a network firewall configuration. While you work in this
area, the configured parameters are saved but not applied (i.e., they are not used to filter packets). The
yellow Yes flag warns you that the configuration has been modified but is not applied.

Managing the Network Firewall Software Configuration Guide
Table 71: Firewall Rule Activation Parameters
Enable This rule is active in the firewall.

Disable This rule is not in the firewall.
Only enabled rules may be applied to the firewall.

3. Enter the source address of the incoming packet in the corresponding Source Address or Interface
field.
Syntax Description
address[/mask] Network IP address (using /mask). The mask must be a plain

number specifying the number of binary 1s at the left side of the
network mask (a mask of 24 specifies a network mask of
255.255.255.0). For instance:
• 192.168.0.11
• 192.168.1.0/24
networkInterfaceName/ The value must already exist in the Interface Configuration table
(see “Interfaces Configuration” on page 100 for more details). The
interface name is case sensitive, hence it must be entered
properly.
is automatically disabled thus removed from the firewall. When
the network interface is enabled or added back, the rule is
automatically enabled and applied in the firewall. For instance:
address.

You can enter a single port or a range of ports. This field supports the following syntax:
port[-port]
any port.
7).
5. Enter the destination address of the incoming packet in the corresponding Destination Address or
Interface field.

Chapter 17 - Network Firewall Configuration Managing the Network Firewall

Syntax Description
address[/mask] Network IP address (using /mask). The mask must be a plain

number specifying the number of binary 1s at the left side of the
network mask (a mask of 24 specifies a network mask of
255.255.255.0). For instance:
• 192.168.0.11
• 192.168.1.0/24
networkInterfaceName/ The value must already exist in the Interface Configuration table
(see “Interfaces Configuration” on page 100 for more details). The
interface name is case sensitive, hence it must be entered
properly.
is automatically disabled thus removed from the firewall. When
the network interface is enabled or added back, the rule is
automatically enabled and applied in the firewall. For instance:
address.

port[-port]
matching any port.
7).
7. Select the protocol of the incoming packet to filter in the corresponding Protocol drop-down menu.
Table 74: Firewall Rule Protocol Parameters

8. Set the corresponding Connection State drop-down menu with the connection state associated with
the incoming packet.
The connection state can be one of the following:
Table 75: Connection State Parameters
State Description
All Match packets in any state.

New Match packets that are not part of an existing connection.

Disabling the Network Firewall Software Configuration Guide
Table 75: Connection State Parameters (Continued)
State Description
Established Or Related Match packets that are part of an existing connection.
9. Select the action to take in the corresponding Action field.

Table 76: Network Firewall Rule Action Parameters

Reject Sends back an ICMP port unreachable in response to the matched packet. The
packet is then dropped.
Note that if a connection is already established before creating a rule that rejects it, this connection
stays active despite the rule applied.
The current enabled rules applied are displayed in the Network > Status web page, which contains
the active configuration in the network firewall. You can also see that the yellow Config Modified
Yes flag is cleared.
Note: You can revert back to the configuration displayed in the Network > Status web page at any time
(including the disabled rules) by clicking the Rollback button at the bottom of the page. All modified settings
in the Network > Network Firewall page will be lost.
Moving a Network Firewall Rule

The firewall rules sequence is very important because only one network firewall rule is applied on a packet.
Rules priority is determined by their position in the table. If you want the unit to try to match one rule before
another one, you must put that rule first.

Deleting a Network Firewall Rule


Disabling the Network Firewall
When the network firewall is enabled, it has an impact on the Aastra unit’s overall performance as the firewall
requires additional processing. You can disable the firewall if you do not need it, thus not impacting
performance. To disable the network firewall, you must stop the NFW service in the System > Services page.
See “Chapter 4 - Services” on page 53 for more details on how to stop a service. All forwarded traffic is allowed
when the network firewall service is stopped.

Chapter 17 - Network Firewall Configuration Disabling the Network Firewall

C H A P T E R
18 NAT Configuration
This chapter describes how to configure the NAT parameters of the Aastra unit.
 Creating/editing a Source NAT
 Creating/editing a Destination NAT
 Moving a NAT rule
 Deleting a NAT rule
Introduction
Network Address Translation (NAT, also known as network masquerading or IP masquerading) rewrites the
source and/or destination addresses/ports of IP packets as they pass through a router or firewall. It is most
commonly used to connect multiple computers to the Internet (or any other IP network) by using one IP
address. This allows home users and small businesses to cheaply and efficiently connect their network to the
Internet. The basic purpose of NAT is to multiplex traffic from the internal network and present it to the Internet
as if it was coming from a single computer having only one IP address.
The Aastra unit’s NAT service allows the dynamic creation and configuration of network address translation
rules. Depending on some criteria, the packet matching the rule may see its source or destination address
modified.
There are two types of NAT rules:
 Source rules: They are applied on the source address of outgoing packets.
 Destination rules: They are applied on the destination address of incoming packets.
A rule's priority is determined by its index in the Source NAT or Destination NAT tables.
If the NAT service is stopped, this tab is greyed out and the parameters are not displayed. See “Chapter 4 -
Services” on page 53 on information on how to start the service.
The maximum number of rules allowed in the configuration is 10 of each Source NAT and Destination NAT.
Caution: Adding source or destination NAT rules has an impact on the Aastra unit’s overall performance
as the NAT requires additional processing. The more rules are enabled, the more overall performance is
affected. Furthermore, Aastra recommends to use a 30 ms packetization time when the NAT is enabled
(instead of a 20 ms ptime, for instance) in order to simultaneously use all the channels available on the unit.
Partial Reset
When a partial reset is triggered, the configuration is rolled back if it was being modified.
A new rule is then automatically applied in the source and in the destination NAT tables to prevent incorrect
rules from blocking access to the unit. If those rules are not the first priority, they are raised. If there are no
rules in the tables, the new rules are not added since there are no rules to override.
Creating/Editing a Source NAT Rule
SNAT rules are executed after the routing decision, before the packet leaves the unit.
The web interface allows you to create a source NAT rule or modify the parameters of an existing one. The
following parameters must all match to apply a SNAT rule to a packet:
 Source Address

Chapter 18 - NAT Configuration Creating/Editing a Source NAT Rule
 Source Port
 Destination Address
 Destination Port
 Protocol
When the above parameters all match, then a new source IP address/port is applied to the packet.
 To create or edit a source NAT rule:

1. In the web interface, click the Network link, then the NAT sub-link.
Figure 48: Source Network Address Translation Rules Section
3 4 5 6 7 8 9
2. In the Source Network Address Translation Rules section of the NAT page, do one of the following:
applied (i.e., the Network Address Translation section of the Status page differs from the NAT page). The
NAT page is a working area where you build up a NAT configuration. While you work in this area, the
configured parameters are saved but not applied (i.e., they are not used in the NAT). The yellow Yes flag
warns you that the configuration has been modified but is not applied.
Table 77: Source NAT Rule Activation Parameters
Enable This SNAT rule is enabled.

Disable This SNAT rule is disabled.
Only enabled rules may be applied to the Source NAT.


Creating/Editing a Source NAT Rule Software Configuration Guide

Syntax Description

the number of binary 1s at the left side of the network mask (a
instance:
• 192.168.0.11
• 192.168.1.0/24
networkInterfaceName[/] The value must already exist in the Interface Configuration table
(see “Interfaces Configuration” on page 100 for more details).
The interface name is case sensitive, hence it must be entered
properly.
automatically enabled and applied in the NAT. For instance:
• Lan1 (Lan1 IP address)

port[-port]
any port.
7).
field.
Table 79: Destination Address Syntax
Syntax Description

the number of binary 1's at the left side of the network mask (a
instance:
• 192.168.0.11
• 192.168.1.0/24

Chapter 18 - NAT Configuration Creating/Editing a Source NAT Rule
Table 79: Destination Address Syntax (Continued)
Syntax Description
networkInterfaceName/ The host address of this interface is used. The value must already
Configuration” on page 100 for more details). The interface name
is case sensitive, hence it must be entered properly.
automatically enabled and applied in the Source NAT. For
instance:
address.

You can enter a single port or a range of ports. This field supports the following format:
port[-port]
matching any port.
7).
8. Select the protocol of the incoming packet to NAT in the corresponding Protocol drop-down menu.
Table 80: Source NAT Rule Protocol Parameters

TCP Matches only TCP packet.
9. Enter the new address applied to the source of the packet in the New Address field.
Table 81: New Address Syntax
Syntax Description
address[:port] Any IP address. When specifying a port number, it is mandatory to

have the protocol set to TCP or UDP.

The current enabled rules applied are displayed in the Network > Status web page, Network
Address Translation section, which contains the active configuration in the NAT. You can also see
that the yellow Config Modified Yes flag is cleared.
Note: You can revert back to the configuration displayed in the Status web page at any time (including the
disabled rules) by clicking the Rollback button at the bottom of the page. All modified settings in the NAT
page will be lost.

Creating/Editing a Destination NAT Rule Software Configuration Guide
Creating/Editing a Destination NAT Rule
The web interface allows you to create a Destination NAT rule or modify the parameters of an existing one.
This creates a rule that allows remote computers (e.g., public machines on the Internet) to connect to a specific
computer within the private LAN, depending on the port used to connect. A destination NAT is also known as
port forwarding or virtual server.
DNAT rules are executed before the routing decision, as the packet enters the unit. Therefore it is important
to configure the Network Firewall (“Chapter 17 - Network Firewall Configuration” on page 135) with respect to
the DNAT rules. An example of this would be port forwarding where the DNAT changes the routed address of
a packet to a new IP address/port. The Network Firewall must also accept connection to this IP/port in order
for the port forwarding to work.
The following parameters must all match to apply a DNAT rule to a packet:
 Source Address
 Source Port
 Destination Address
 Destination Port
 Protocol
When the above parameters all match, then a new destination IP address/port is applied to the packet.
 To create or edit a Destination NAT rule:

1. In the Destination Network Address Translation Rules section of the NAT page, do one of the
following:
Figure 49: Destination Network Address Translation Rules Section

2 3 4 5 6 7 8
Note: When you add a new rule, edit an existing rule, or delete a rule, you can see a yellow Yes in the
Config Modified section at the top of the window. It warns you that the configuration has been modified but
not applied (i.e., the Network Address Translation section of the Status page differs from the NAT page).
The NAT page is a working area where you build up a NAT configuration. While you work in this area, the
configured parameters are saved but not applied (i.e., they are not used in the NAT). The yellow Yes flag
warns you that the configuration has been modified but is not applied.
Table 82: Destination NAT Rule Activation Parameters
Enable This DNAT rule is enabled.

Disable This DNAT rule is disabled.
Only enabled rules may be applied to the Destination NAT.


Chapter 18 - NAT Configuration Creating/Editing a Destination NAT Rule

Syntax Description

instance:
• 192.168.0.11
• 192.168.1.0/24
networkInterfaceName/ The host address of this interface is used. The value must already
automatically enabled and applied in the Destination NAT. For
instance:
address.

port[-port]
any port.
7).
field.
Table 84: Destination Address Syntax
Syntax Description

instance:
• 192.168.0.11
• 192.168.1.0/24

Creating/Editing a Destination NAT Rule Software Configuration Guide
Table 84: Destination Address Syntax (Continued)
Syntax Description
networkInterfaceName[/] The host address of this interface is used. The value must already
automatically enabled and applied in the Destination NAT. For
instance:
• Lan1 (Lan1 IP address)

port[-port]
matching any port.
7).
7. Select the protocol of the incoming packet to NAT in the corresponding Protocol drop-down menu.
Table 85: Destination NAT Rule Protocol Parameters

8. Enter the new address of the packet in the New Address field.
Table 86: New Address Syntax
Syntax Description
address[:port] Any IP address. When specifying a port number, it is mandatory to

have the protocol set to TCP or UDP.

The current enabled rules applied are displayed in the Network > Status web page, Network
Address Translation section, which contains the active configuration in the NAT. You can also see
Note: You can revert back to the configuration displayed in the Network > Status web page at any time
(including the disabled rules) by clicking the Rollback button at the bottom of the page. All modified settings
in the Network > NAT page will be lost.

Chapter 18 - NAT Configuration Moving a NAT Rule
Moving a NAT Rule
The NAT rules sequence is very important because only one SNAT rule or one DNAT rule is applied on a
packet. Rules priority is determined by their position in the table. If you want the unit to try to match one rule
before another one, you must put that rule first.

Deleting a NAT Rule

Disabling the NAT
When the NAT is enabled, it has an impact on the Aastra unit’s overall performance as the NAT requires
additional processing. You can disable the NAT if you do not need it, thus not impacting performance. To
disable the NAT, you must stop the NAT service in the System > Services page. See “Chapter 4 - Services”
on page 53 for more details on how to stop a service.

C H A P T E R
19 DHCP Server Settings
This chapter describes how to configure the embedded DHCP server of the Aastra unit.
Standards Supported • RFC 2131: Dynamic Host Configuration Protocol, section 2

(server side)
• RFC 2132: DHCP Options and BOOTP Vendor Extensions
(sections 3.3, 3.5, 3.8, 3.17, 8.3 and 8.5)
Introduction
The Aastra unit contains an embedded DHCP server that allocates IP addresses and provides leases to the
various subnets that are configured. These subnets could have PCs or other IP devices connected to the unit’s
LAN Ethernet connectors. These devices could be any combination of switches, PCs, IP phones, etc.
If the DHCP service is stopped, this tab is greyed out and the parameters are not displayed. See “Chapter 4 -
Services” on page 53 on information on how to start the service.
Note: The Aastra unit’s DHCP server settings do not support IPv6. See “IPv4 vs. IPv6” on page 85 for more
details.
Subnet Server
The DHCP server manages the hosts’ network configuration on a given subnet. Each subnet can be seen as
having a distinct DHCP server managing it, which is called a subnet server. To activate a subnet server for a
given network interface, the name of that network interface and the name of the subnet configuration must
match (the names are case sensitive). Only one subnet can be defined per network interface. The network
interface can be a physical interface or a logical interface (e.g., sub-interface using VLAN).
Leases
In order to assign leases, the subnet server draws from an IP address pool (or subnet scope) defined by a
start address and an end address. The subnet mask assigned to hosts is taken directly from the network
interface. All hosts on the same subnet share the same configuration. The maximum number of hosts
supported on a subnet is 254.
You can reserve IP addresses for specific hosts that are designated by their MAC address. Those addresses
are then removed from the pool of IP addresses that can be leased. Once a lease is assigned, it is removed
from the pool of IP addresses that can be leased for as long as the host keeps it.
Configuration Parameters
When an address is leased to a host, several network configuration parameters are sent to that host at the
same time according to the options found in the DHCP request. You can modify the configuration source of a
parameter. The following are the possible configuration sources:
Table 87: Parameter Configuration Sources
Source Description
Static The parameter is defined as a static parameter locally.

Automatic The parameter is obtained from the network configured in the Automatic Configuration
Interface drop-down menu of this subnet (“DHCP Basic Configuration” on page 137).

Chapter 19 - DHCP Server Settings Introduction
Table 87: Parameter Configuration Sources (Continued)
Source Description
Host Configuration The parameter is obtained from the host configuration.

Host Interface The parameter is obtained from the network interface matching the subnet.
The following table lists the configuration parameters and their available configuration sources:
Table 88: Optional Parameter and Possible Configuration Sources
Configuration Sources
Parameter Name
Static Automatic Host Config Host Interface
Domain Name
Lease time
Default gateway
List of DNS servers
List of NTP servers
List of NBNS servers
Default vs. Specific Configurations

You can use two types of configuration:
 Default configurations that apply to all the subnets of the Aastra unit.
 Specific configurations that override the default configurations.
You can define specific configurations for each subnet in your Aastra unit. For instance, you could
define a lease time for all the subnets of the Aastra unit and use the specific configuration
parameters to set a different value for one specific subnet.
The parameters available differ according to the subnet you have selected. The Default subnet has less
parameters than the specific subnets available on the Aastra unit.

DHCP Basic Configuration Software Configuration Guide
DHCP Basic Configuration
The basic configuration parameters are available only on the specific subnets configuration.
 To set the DHCP server basic parameters:

1. In the web interface, click the Network link, then the DHCP Server sub-link.
2. Select a specific subnet in the Select Subnet drop-down menu at the top of the window.
You have the choice between Default (applies to all subnets) and specific subnets.
3. In the DHCP Server Configuration section of the DHCP Server page, enable the DHCP server by
selecting Enable in the DHCP Server Enable drop-down menu.
Figure 50: DHCP Server Configuration – General Parameters
3
4
5
4. Set the start and end IP addresses of the subnet range in the Start IP Address and End IP Address
fields.
These are the addresses that the DHCP server offers to the subnets of the Aastra unit. The Aastra
unit can offer up to 254 addresses. These addresses must be within the network interface’s subnet
or the subnet server will have an invalid configuration status.
5. Set the Automatic Configuration Interface drop-down menu with the network interface that provides
the automatic configuration (e.g.: DNS servers, NTP server, etc.) to all parameters of this subnet
that use the "Automatic" configuration source.
Lease Time (Option 51)
The Aastra unit DHCP server offers a lease time to its subnets. You can use a default lease time for all subnets
or define a lease time specific to one or more subnets.
 To set the DHCP server lease time parameters:

1. In the Lease Time (Option 51) sub-section of the DHCP Server Configuration section, define
whether or not you want to override the lease time set in the Default configuration in the Subnet
Specific drop-down menu.
This menu is available only in the specific subnets configuration.
Figure 51: DHCP Server Configuration – Lease Time Option
1
2
2. Define the lease time (in seconds) given by the Aastra unit DHCP server in the Lease Time field.

Chapter 19 - DHCP Server Settings Domain Name (Option 15)
Domain Name (Option 15)
The Aastra unit DHCP server offers a domain name to its subnets. You can use a default domain name for all
subnets or define a domain name specific to one or more subnets.
 To set the DHCP server domain name parameters:

1. In the Domain Name (Option 15) sub-section of the DHCP Server Configuration section, enable the
domain name (option 15) by selecting Enable in the Enable Option drop-down menu.
Figure 52: DHCP Server Configuration – Domain Name Option
1
2
3
4
2. Define whether or not you want to override the domain name parameters set in the Default
configuration in the Subnet Specific Value drop-down menu.
3. If the domain name option is enabled, select the configuration source of the domain name
Table 89: Domain Name Configuration Sources
Source Description
Host The domain name is the one used by the unit.

Configuration
Static You manually enter a domain name.
4. If the configuration source is Static, enter the static default domain name for all subnets in the
Domain Name field.
Default Gateway (Option 3)
The Aastra unit DHCP server offers a default gateway (also called default router) to its subnets.
Note: The default gateway parameters are not available in the Default interface. You must access the
specific subnets configuration to set its parameters.
 To set the DHCP server default gateway parameters:

1. In the Default Gateway (Option 3) sub-section of the DHCP Server Configuration section, enable
the default gateway (option 3) by selecting Enable in the Enable Option drop-down menu
Figure 53: DHCP Server Configuration – Default Gateway Option
1
2
3

DNS (Option 6) Software Configuration Guide
2. Select the configuration source of the default gateway information in the Configuration Source drop-
down menu.
Table 90: Default Gateway Configuration Sources
Source Description
Host Interface The default gateway is the host address within the client's subnet.
Static You manually enter the value.
3. If the configuration source is Static, enter the default gateway host name or IP address of the
subnet in the Default Gateway field.
DNS (Option 6)
The Aastra unit DHCP server offers up to four DNS addresses to its subnets. You can use the default DNS
addresses for all subnets or define static DNS addresses specific to one or more subnets.
 To set the DHCP server DNS parameters:

1. In the DNS (Option 6) sub-section of the DHCP Server Configuration section, enable the DNS
servers (option 6) by selecting Enable in the Enable Option drop-down menu
Figure 54: DHCP Server Configuration – DNS Option
1
2
3
2. Define whether or not you want to override the default values in the Subnet Specific drop-down
menu.
3. Select the configuration source of the DNS information in the Configuration Source drop-down
menu.
Table 91: DNS Configuration Sources
Source Description
Host The DNS servers are obtained from the host configuration.
Configuration
Automatic The DNS servers are automatically obtained from the network configured in the
Automatic Configuration Interface drop-down menu of this subnet (“DHCP
Basic Configuration” on page 137).
4. If the configuration source is Static, enter the static addresses of up to four DNS servers in the
following fields:

Chapter 19 - DHCP Server Settings NTP (Option 42)
• Primary DNS
• Secondary DNS
• Third DNS
• Fourth DNS
NTP (Option 42)
The Aastra unit DHCP server offers the addresses of up to four NTP (Network Time Protocol) servers to its
subnets. You can use the default NTP addresses for all subnets or define static DNS addresses specific to
one or more subnets.
 To set the DHCP server NTP parameters:

1. In the NTP (Option 42) sub-section of the DHCP Server Configuration section, enable the NTP
Figure 55: DHCP Server Configuration – NTP Option
1
2
3
menu.
3. Select the configuration source of the NTP information in the Configuration Source drop-down
menu.
Table 92: NTP Configuration Sources
Source Description
Host The NTP servers are obtained from the host configuration.
Configuration
Automatic The NTP servers are automatically obtained from the network configured in the
Automatic Configuration Interface drop-down menu of this subnet (“DHCP
Basic Configuration” on page 137).
4. If the configuration source is Static, enter the static addresses of up to four NTP servers in the
following fields:
• Primary NTP
• Secondary NTP
• Third NTP
• Fourth NTP

NBNS (Option 44) Software Configuration Guide
NBNS (Option 44)
The NetBIOS Name Server (NBNS) protocol, part of the NetBIOS over TCP/IP (NBT) family of protocols, is
implemented in Windows systems as the Windows Internet Name Service (WINS). By design, NBNS allows
network peers to assist in managing name conflicts.
The Aastra unit DHCP server offers up to four NBNS addresses to its subnets. You can use the default NBNS
addresses for all subnets or define static NBNS addresses specific to one or more subnets.
 To set the DHCP server NBNS parameters:

1. In the NBNS (Option 44) sub-section of the DHCP Server Configuration section, enable the NBNS
Figure 56: DHCP Server – NBNS Option
1
2
menu.
3. Enter the static addresses of up to four NBNS servers in the following fields:
• Primary NBNS
• Secondary NBNS
• Third NBNS
• Fourth NBNS
DHCP Static Leases Configuration
The embedded DHCP server leases addresses to the hosts that request it. The address is assigned to a host
for a configurable amount of time (as defined in “Lease Time (Option 51)” on page 137). The DHCP server
can service all subnets on which it is enabled.
 To define DHCP leases offered by the Aastra unit:

1. In the web interface, click the System link, then the DHCP Leases sub-link.
2. In the Static Leases Configuration section, if applicable, delete an existing reserved IP address by
selecting Delete in the Action drop-down next to an existing lease.
3. If applicable, add a new lease by entering the MAC address of the device and the IP address you
want to reserve for it, then click Submit.

Chapter 19 - DHCP Server Settings DHCP Static Leases Configuration
The static IP address is added to the Static Leases Configuration section, but not to the Current
Leases section.

POTS Parameters
C H A P T E R
20 POTS Configuration
This chapter describes how to configure the POTS (Plain Old Telephony System) line service, which allows
you to configure the analog specification of each line, as well as gateways-specific parameters.
POTS Status
The POTS parameters are displayed in the POTS / Status page.
Line Status
The Line Status table lists the link state of the FXS lines.
Figure 57: POTS – Status Web Page
The State column may have one of the following values:

 Idle: The line is available
 In Use: The line is currently used
 Disabled: The line is disabled
 Bypass: The line is on bypass
 Down: The power of the line is down
General POTS Configuration
The General Configuration section allows you to select the detection/generation method of caller ID.

Chapter 20 - POTS Configuration General POTS Configuration
 To configure the general POTS parameters:

1. In the web interface, click the POTS link, then the Config sub-link.
Figure 58: POTS Web Page
2
3
4
2. Select the detection/generation method of caller ID in the Caller ID customization drop-down menu.
This allows selecting the detection/generation method of caller ID. See “Caller ID Information” on
Table 93: Caller ID Parameters
Country Uses the default caller ID of the country defined in the Country section of the
Telephony > Misc page (“Country Configuration” on page 451).
EtsiDtmf ETSI 300 659-1 (DTMF string sent between the first and second ring).
EtsiFsk ETSI 300 659-1 (FSK (V.21) sent between the first and second ring).
3. Select the caller ID transmission method in the Caller ID Transmission drop-down menu.
It allows selecting the transmission type of the caller ID.
Table 94: Caller ID Transmission Parameters
Country Uses the default caller ID of the country defined in the Country section of the
Telephony > Misc page (“Country Configuration” on page 451).
First Ring The caller ID is sent after the first ring.
Ring Pulse The caller ID is sent between a brief ring pulse and the first ring.
Line Reversal The caller ID is sent between a brief ring pulse and the first ring on an inverted
Ring Pulse polarity line.
DT-AS The caller ID is sent after the dual tone alerting state tone.
Line Reversal The caller ID is sent after the dual tone alerting state tone on an inverted
DT-AS polarity line.
No Ring Pulse The caller ID is sent before the first ring.
4. Determine the type of vocal information that can be obtained by dialing a pre-defined digit map in
the Vocal Unit Information drop-down menu.
When entering special characters on your telephone pad, the Aastra unit talks back to you with
relevant information.
Table 95: Caller ID Parameters
None The vocal information feature is disabled.

General POTS Configuration Software Configuration Guide
Table 95: Caller ID Parameters (Continued)
All Enable all vocal information digit maps.
To access the vocal unit information:

a. Take one of the telephones connected to the Aastra unit.
b. Dial one of the digits sequence on the keypad.
Table 96: Vocal Unit Information
Digits to Dial Information Vocally Sent by the Aastra unit
*#*0 List of IP addresses of the Aastra unit (static or DHCP).

*#*1 MAC address of the Aastra unit.
*#*8 Firmware version number of the Aastra unit.
Caller ID Information
The caller ID is a generic name for the service provided by telephone utilities that supply information such as
the telephone number or the name of the calling party to the called subscriber at the start of a call. In call
waiting, the caller ID service supplies information about a second incoming caller to a subscriber already busy
with a phone call. However, note that caller ID on call waiting is not supported by all caller ID-capable
telephone displays.
In typical caller ID systems, the coded calling number information is sent from the central exchange to the
called telephone. This information can be shown on a display of the subscriber telephone set. In this case, the
caller ID information is usually displayed before the subscriber decides to answer the incoming call. If the line
is connected to a computer, caller information can be used to search in databases and additional services can
be offered.
The following basic caller ID features are supported:
 Date and Time
 Calling Line Identity
 Calling Party Name
 Visual Indicator (MWI)
Caller ID Generation
There are two methods used for sending caller ID information depending on the application and country-
specific requirements:
 caller ID generation using DTMF signalling
 caller ID generation using Frequency Shift Keying (FSK)
Note: The Dgw v2.0 Application does not support ASCII special characters higher than 127.
The displayed caller ID for all countries may be up to 20 digits for numbers and 50 digits for names.
DTMF Signalling
The data transmission using DTMF signalling is performed during or before ringing depending on the country
settings or endpoint configuration. The Aastra unit provides the calling line identity according to the following
standards:
 Europe: ETSI 300 659-1 January 2001 (Annex B): Access and Terminals (AT); Analogue
access to the Public Switched Telephone Network (PSTN); Subscriber line protocol over the

Chapter 20 - POTS Configuration FXS Configuration
local loop for display (and related) services; Part 1: On-hook data transmission.
FSK Generation
Different countries use different standards to send caller ID information. The Aastra unit is compatible with the
following widely used standards:
 ETSI 300 659-1
Note: The compatibility of the Aastra unit is not limited to the above caller ID standards.
Continuous phase binary FSK modulation is used for coding that is compatible with:
 BELL 202
 ITU-T V.23
FXS Configuration
The FXS Configuration section allows you to define how a FXS endpoint behaves in certain conditions.
 To configure the FXS parameters:

1. In the web interface, click the POTS link, then the FXS Config sub-link.
Figure 59: FXS Config Web Page
2
3
4
5
6
7
8
2. In the FXS Configuration section, set the Line Supervision Mode drop-down menu with the power
drop and line polarity used to signal the state of a line.
Power drop and polarity reversal are also called battery drop and battery reversal.
Table 97: Line Supervision Mode Parameters
None Power drop or polarity reversal is not used to signal the state of the
line.
DropOnDisconnect Activates the Power Drop on Disconnect feature. A short power drop
is made at the end of a call when the call is disconnected by the
remote party.
The drop duration can be configured in the FXS Power Drop on
Disconnect Duration field (Step 5).

FXS Configuration Software Configuration Guide
Table 97: Line Supervision Mode Parameters (Continued)
ReversalOnIdle Activates the Polarity Reversal on Idle feature. The polarity of the line
is initially in reversed state. The polarity of the line returns to the
positive state when the user seizes the line or when the line rings for
an incoming call. The polarity of the line is reversed again when the
call is disconnected.
ReversalOnEstablished Activates the Polarity Reversal on Established option. The polarity of
the line is initially in the positive state. The polarity of the line is
reversed when the call is established and returns to the positive state
when the call is disconnected.
3. Set the Disconnect Delay field with the value used to determine whether or not call clearing occurs
as soon as the called user is the first to hang up a received call.
This parameter has no effect when you are acting as the calling party.
If you set the value to 0, the call is disconnected as soon as the called user hangs up the call.
If the value is greater than 0, that value is the amount of time, in seconds, the unit waits after the
called user hangs up before signalling the end of the call.
4. Set the Auto Cancel Timeout field with the time, in seconds, the endpoint rings before the call is
automatically cancelled.
Setting this variable to 0 disables the timeout. Calls will not be automatically cancelled and will ring
until the party answers.
5. Set the Inband Ringback drop-down menu to define whether or not the FXS endpoint needs to
generate a ringback for incoming ringing call.
Table 98: Inband Ringback Parameters
Disable The FXS endpoint does not play local ringback to the remote party.
Enable The FXS endpoint plays local ringback to the remote party via the negotiated
media stream. The local ringback is generated only when the telephone is on-
hook. The FXS ports never play the local ringback for the call waiting.
6. Set the Shutdown Behavior drop-down menu with the FXS endpoint behavior when it becomes shut
down.
Table 99: FXS Shutdown Behavior Parameters
Disabled A disabled tone is played when the user picks up the telephone and the FXS
Tone endpoint is shut down.
Power The loop current is interrupted when the FXS endpoint is shut down and no tone is
Drop played when the user picks up the telephone.
A FXS endpoint becomes shut down when the operational state of the endpoint becomes Disabled
and the Shutdown Endpoint When Operational State is 'Disable' And Its Usage State Is 'idle-
unusable' parameter of the SIP > Endpoints page is set to Enable. See “Administration” on page 68
for more details.
This parameter is not used by FXS endpoints used for bypass when the Activation column of the
FXS Bypass section is set to Endpoint Disabled. See “FXS Bypass” on page 167 for more details.
7. Set the Power Drop on Disconnect Duration field with the power drop duration, in milliseconds, that
is made at the end of a call when the call is disconnected by the remote party.

Chapter 20 - POTS Configuration FXS Country Customization
This value only has an effect when the Line Supervision Mode drop-down menu is set to
DropOnDisconnect.
8. Set the Service Activation drop-down menu with the method used by the user to activate
supplementary services such as call hold, second call, call waiting, call transfer and conference call.
Table 100: Service Activation Parameters
Flash Hook Service activation is performed by flash hook or hanging up.

Flash Hook And Service activation is performed by flash hook, flash hook followed by a digit
Digit or hanging up.
The digit dialed has a different behaviour depending on the current call
context:
• One call active and one waiting call:
Flash hook then dial the digit 2: Answer the waiting call.
• One call active and one call on hold:
Flash hook then dial the digit 1: Terminate the active call and recover
the call on hold.
Flash hook then dial the digit 2: Hold the active call and recover the
call on hold.
Flash hook then dial the digit 3: Enter the conference mode.
Flash hook then dial the digit 4: Transfer the call on hold to the active
call.
When hanging up in this context, the telephone rings to notify the user
there is still a call on hold.
• In conference mode:
Flash hook then dial the digit 2: Return to one active call and one call
on hold.
When hanging up in this context, all calls are finished.
FXS Country Customization
The FXS Country Customization section allows you to override the current default country parameters of
certain features. Refer to “Appendix A - Country-Specific Parameters” on page 603 for the pre-defined values
for a specific country.
 To define the FXS country customization parameters:

1. In the FXS Country Configuration section, select whether or not you want to override the current
country parameters in the Override Country Customization drop down menu.
This allows overriding FXS related default country settings for the loop current and flash hook
detection features.

FXS Country Customization Software Configuration Guide
Figure 60: FXS Country Customization Section
1
2
3
Table 101: Line Supervision Mode Parameters
Disable The line uses the default country FXS settings.

Enable The line uses the FXS country configuration set in the following
steps.
2. Set the Country Override Loop Current field with the loop current generated by the FXS port in ma.
When a remote end-user goes on-hook, the Aastra unit signals the far end disconnect by performing
a current loop drop (< 1 mA) on the analog line. This current loop drop, also referred to as “Power
Denial” mode, is typically used for disconnect supervision on analog lines. The Aastra unit maintains
a current drop for one second (this value cannot be configured), then a busy tone is generated to
indicate the user to hang up. See the description for the FXS Line Supervision Mode drop-down
menu in “FXS Configuration” on page 148 for more details.
When one of its analog lines goes off-hook, the Aastra unit controls the endpoint in a fixed loop
current mode. When selecting a country (see “Country Configuration” on page 451 for more details),
each country has a default loop current value. However, you can override this value and define your
own loop current.
Note that the actual measured current may be different than the value you set, because it varies
depending on the DC impedance.
3. Set the Country Override Flash Hook Detection Range field.
This is the range in which the hook switch must remain pressed to perform a flash hook.
When selecting a country (see “Country Configuration” on page 451 for more details), each country
has a default minimum and maximum time value. However, you can override these values and
define your own minimum and maximum time within which pressing and releasing the plunger is
actually considered a flash hook.
The range consists of the minimal delay and maximal delay, in ms, separated by a “-”. The minimal
value allowed is 10 ms and the maximum value allowed is 1200 ms. The space character is not
allowed.
Flash hook can be described as quickly depressing and releasing the plunger in or the actual
handset-cradle to create a signal indicating a change in the current telephone session. Services
such as picking up a call waiting, second call, call on hold, and conference are triggered by the use
of the flash hook.
A flash hook is detected when the hook switch is pressed for a shorter time than would be required
to be interpreted as a hang-up.
Using the “flash” button that is present on many standard telephone handsets can also trigger a
flash hook.
Calling Party Name of the Caller ID

Standards Supported • ETSI EN 300659-3a
a. CLIR section
1. In the potsMIB, specify the Calling Party Name of the caller ID (CLIP) when the calling party is
tagged as private in the FxsCallerIdPrivateCallingPartyName variable.

Chapter 20 - POTS Configuration FXS Country Customization
pots.FxsCallerIdPrivateCallingPartyName="Value"
Value may be any string of characters up to 50 characters.

• When empty, no Calling Party Name parameter is sent.
• When set to 'P', no Calling Party Name parameter is sent but a Reason for Absence or
Caller Party Name parameter is sent with the value 0x50 (Private).

SIP Parameters
C H A P T E R
21 SIP Gateways
This chapter describes how to add and remove SIP gateways in the Aastra unit.
SIP Gateways Configuration
Multiple SIP gateways may be used for a number of reasons, such as:
 Redirecting ISDN calls to different SIP servers depending on the call.
 Hunt calls across several gateways.
Adding a SIP gateway triggers a warning message if the total number of registrations configured reached the
defined limit. See “Number of Registrations” on page 293 for more details.
 To configure multiple SIP gateways:

1. In the web interface, click the SIP link, then the Gateways sub-link.
Figure 61: SIP – Gateways Web Page
2 3 4 5 6
You can add a new gateway by clicking the button. The Aastra unit supports a maximum of 5
gateways.
You can delete an existing gateway by clicking the button.
2. If you are adding a new gateway, enter its name in the Name field.
The Dgw v2.0 Application supports only alphanumeric characters, “-”, and “_”.
3. Select the network interface on which the gateway listens for incoming SIP traffic in the Signaling
Network drop-down menu.
This value applies to all transports (e.g., UDP, TCP, etc.).
The LAN interface may be used as a SIP gateway to be bound on the LAN. However, there is no
routing between the LAN and the uplink interface.
4. Define the list of networks (separated by ",") to use for the media (voice, fax, etc.) stream in the
Media Networks field.
You can use the Media Networks Suggestion column’s drop-down menu to select between
suggested values, if any.
The value must match one of the "InterarfaceName" values in the "NetworkInterfacesStatus" table
of the BNI service. The order in the list defines the priority.
When the media stream is negotiated, the following rules apply:

Chapter 21 - SIP Gateways SIP Gateways Configuration
• If the list of media networks is empty,the Aastra unit uses the IP address of the network
defined in the Signaling Network drop-down menu.
• Only active networks are used.
• Only the first active network of an IP address family (IPv4, IPv6) is used. All
subsequent networks of the same IP family are ignored.
Note: When generating an offer and multiple networks are available for the media, ANAT grouping (RFC
4091) is automatically enabled. When generating an answer, the ANAT grouping state is detected form the
offer.
5. Set the SIP port on which the gateway listens for incoming unsecure SIP traffic in the Port field.
This is used only when the UDP and/or TCP transports are enabled.
If two or more SIP gateways use the same port, only the first SIP gateway starts correctly. The
others are in error and not started. The SIP gateway is also in error and not started if the port is
already used.
The default value is 0. If you set the port to 0, the default SIP port 5060 is used.
Note: The port “0” is the equivalent to the “well known port”, which is 5060 in SIP. Using 0 and 5060 is not
the same. At the SIP packets level, if you set the port to 0, it will not be present in the SIP packet. If you set
the port to 5060, it will be present in the SIP packet. For example: “[email protected]” if the port is 0 and
“[email protected]:5060” if the port is 5060.
6. Set the SIP port on which the gateway listens for incoming secure SIP traffic in the Secure Port field.
This is used only when the TLS transport is enabled.
The default value is 0. If you set the port to 0, the default secure SIP port 5061 is used.
Note: The port “0” is the equivalent to the “well known port”, which is 5061 in SIP for TLS. Using 0 and 5061
is not the same. At the SIP packets level, if you set the port to 0, it will not be present in the SIP packet. If
you set the port to 5061, it will be present in the SIP packet. For example: “[email protected]” if the port is 0 and
“[email protected]:5061” if the port is 5061.

The state of the SIP gateways is displayed in the SIP Gateway Status section.
Table 102: SIP Gateway States
State Description
Ready The gateway is ready to make and receive calls.

Cannot start, port already in The gateway cannot open its IP port because the port is
use already used by another service. This generally occurs when
the administrator adds a new gateway but forgets to configure
a different IP port.
Network down The SIP gateway is not started or the network interface on
which the SIP gateway is associated does not have an IP
address.
Restarting The SIP gateway cannot make or receive calls while it is
restarting.
Waiting for time The gateway is started but it cannot open its SIP TLS port
synchronization because the real-time clock is not synchronized. This generally
occurs when the SNTP server is not set or is unreachable.
Server unreachable The gateway is started but it cannot make and receive calls
because the SIP server is unreachable. This state is only
reported when a KeepAlive mechanism is used.

SIP Gateways Configuration Software Configuration Guide
Table 102: SIP Gateway States (Continued)
State Description
Unregistered Indicates some registrations that are mandatory for this

gateway failed. See “Unregistered User Behaviour” on

Chapter 21 - SIP Gateways SIP Gateways Configuration

C H A P T E R
22 SIP Servers
This chapter describes how to configure the SIP server parameters of the Aastra unit.
Standards Supported • RFC 2543: SIP: Session Initiation Protocol

• RFC 3261: The Session Initiation Protocol (SIP)
• RFC 3903: Session Initiation Protocol (SIP) Extension for
Event State Publication
It describes the following:

 How to define the SIP servers IP information.
 How to define the SIP gateways IP information.
Introduction
The Aastra unit uses the following types of servers:

Table 103: SIP Servers
Server Description
Registrar Server Accepts REGISTER requests and places the information it receives in those
requests into the location service for the domain it handles.
Proxy Server An intermediary program that acts as both a server and a client for the purpose of
making requests on behalf of other clients. A proxy server primarily plays the role
of routing, which means its job is to ensure that a request is passed on to another
entity that can further process the request. Proxies are also useful for enforcing
policy and for firewall traversal. A proxy interprets, and, if necessary, rewrites
parts of a request message before forwarding it.
Outbound Proxy Server An intermediary entity that acts as both a server and a client for the purpose of
making requests on behalf of other clients. The outbound proxy receives all
outbound traffic and forwards it. Incoming traffic may or may not go through the
outbound proxy. The outbound proxy’s address is never used in the SIP packets,
it is only used as a physical network destination for the packets.
When the outbound proxy is enabled, the proxy is still used to create the To and
From headers, but the packets are physically sent to the outbound proxy.
Messaging Server Host A Messaging system host is a server that accepts MWI SUBSCRIBE requests
and places the information it receives in those requests into the location service
for the domain it handles.
SIP Outbound Proxy (From RFC 3261)

A proxy that receives requests from a client, even though it may not be the server resolved by the
Request-URI. Typically, a user agent is manually configured with an outbound proxy.
When enabled, the initial route for all SIP requests contains the outbound proxy address, suffixed with
the loose routing parameter “lr”. The Request-URI still contains the home domain proxy address.
Requests are directed to the first route (the outbound proxy).

Chapter 22 - SIP Servers TLS Persistent Connections Status
TLS Persistent Connections Status
The TLS Persistent Connections Status table allows you to browse the status of the TLS persistent
connections of the Aastra unit. These connections are associated with the SIP servers (outbound proxy,
registrar and home domain proxy). Note that this section is not displayed if there is no information to show.
Figure 62: SIP – TLS Persistent Connections Status Section
The following information is available:

Table 104: TLS Persistent Connection Parameters
Gateway The SIP gateway used to register.

Local Port Local port used by the TLS persistent connection.
Remote Host The remote host used to establish the TLS persistent connection. The remote host
can be a host name or an IP address of the proxy, outbound proxy or registrar.
Remote IP Address The resolved IP address of the remote host used to establish the TLS persistent
connection.
Status The current state of the TLS persistent connection.
• Up: The TLS connection is established.
• Down: The TLS connection is not established.
SIP Servers Configuration
This section describes how to configure the IP address and port number of the SIP servers.
If any of the SIP servers parameters corresponds to a domain name that is bound to a SRV record, the
corresponding port must be set to 0 for the unit to perform DNS requests of type SRV (as per RFC 3263).
Otherwise, the unit will not use DNS SRV requests, but will rather use only requests of type A because it does
not need to be specified which port to use.

Multiple SIP Gateways Software Configuration Guide
 To set the SIP servers configuration:

1. In the web interface, click the SIP link, then the Configuration sub-link.
Figure 63: SIP – Servers Web Page
2
3
4
5
2. Enter the SIP registrar server static IP address or domain name and port number in the Registrar
Host field.
You must enter the information as IP address:Port number. For instance:
192.168.0.5:5060
3. Enter the SIP Proxy server static IP address or domain name and port number in the Proxy Host
field.
192.168.0.5:5060
4. Enter the SIP outbound proxy server static IP address or domain name and port number in the
Outbound Proxy Host field.
The outbound proxy is enabled if the IP address is valid (i.e., not 0.0.0.0:0). Setting the address to
0.0.0.0:0 or leaving the field empty disables the outbound proxy.
5. Enter the Messaging system host static IP address or domain name and port number in the
Messaging Server Host field.
If the host corresponds to a domain name that is bound to a SRV record, the port must be set to 0
for the unit to perform DNS SRV queries; otherwise only type A record lookups will be used.
You can define whether or not an endpoint needs to subscribe to a messaging system in “Endpoints
Registration” on page 289.
Multiple SIP Gateways
The Aastra unit allows you to have multiple SIP gateways (interfaces). You can configure each SIP gateway
to register to a specific registrar. You can also configure each SIP gateway to send all requests to an outbound
proxy. See “Chapter 24 - SIP Gateways” on page 277 for more details.
SIP Gateway Specific Registrar Servers

This section allows you to define whether the available SIP gateways use the default registrar server or rather
use a specific registrar server.
 To set specific registrars servers information:

1. In the Registrar Servers section of the Servers page, select whether or not a SIP gateway uses a
specific registrar server in the Gateway Specific drop-down menu.
If you select No, the SIP gateway uses the server information as set in the SIP Default Servers
section.

Chapter 22 - SIP Servers Multiple SIP Gateways
Figure 64: SIP Servers – Specific Registrar Section

1 2
2. Enter the IP address or domain name and port number of the registrar server currently used by the
registration in the Registrar Host field.
192.168.0.5:5060
3. If you do not need to set other parameters, do one of the following:

• To save your settings without refreshing the registration, click Submit.
• To save your settings and refresh the registration now, click Submit & Refresh
Registration.
SIP Gateway Specific Messaging Servers

This section allows you to define whether the available SIP gateways use the default proxy and outbound
proxy server or rather use specific servers.
 To set specific proxy servers information:

1. In the Messaging Servers section of the Servers page, select whether or not a SIP gateway uses a
specific proxy and outbound proxy server in the Gateway Specific drop-down menu.
If you select No, the SIP gateway uses the server information as set in the SIP Default Servers and
Messaging Subscription (“Messaging Subscription” on page 349) sections.
Figure 65: SIP Servers – Messaging Section

1 2
2. Enter the IP address or domain name and port number of the messaging server currently used by
the registration in the Proxy Host field.
192.168.0.5:5060
3. Enter the IP address or domain name and port number of the outbound proxy server currently used
by the registration in the Outbound Proxy Host field.
192.168.0.5:5060
Registration.
SIP Gateway Specific Proxy Servers

This section allows you to define whether the available SIP gateways use the default proxy and outbound
proxy server or rather use specific servers.

Multiple SIP Gateways Software Configuration Guide
 To set specific proxy servers information:

1. In the Proxy Servers section of the Servers page, select whether or not a SIP gateway uses a
specific proxy and outbound proxy server in the Gateway Specific drop-down menu.
If you select No, the SIP gateway uses the server information as set in the SIP Default Servers
section.
Figure 66: SIP Servers – Specific Proxy Section

1 2 3
2. Enter the IP address or domain name and port number of the proxy server currently used by the
registration in the Proxy Host field.
192.168.0.5:5060
3. Enter the IP address or domain name and port number of the outbound proxy server currently used
by the registration in the Outbound Proxy Host field.
192.168.0.5:5060
Registration.
Keep Alive
You can select the method used to perform the SIP keep alive mechanism. With this mechanism, the Aastra
unit sends messages periodically to the server to ensure that it can still be reached.
 To use the SIP keep alive mechanism:

1. In the Keep Alive section of the Servers page, select the keep alive method to use in the Keep Alive
Method drop-down menu.
Figure 67: Keep Alive Section
1
2
3
Table 105: Keep Alive Parameters
None No keep alive is performed.

Chapter 22 - SIP Servers Multiple SIP Gateways
Table 105: Keep Alive Parameters (Continued)
SipOptions SIP OPTIONS are sent periodically for each gateway to the
corresponding server. Any response received from the
server means that it can be reached. No additional
processing is performed on the response. If no response is
received after the retransmission timer expires
(configurable via the Transmission Timeout field in “SIP
Interop” on page 312), the gateway considers the server as
unreachable. In this case, any call attempt through the
gateway is refused. SIP OPTIONS are still sent when the
server cannot be reached and as soon as it can be reached
again, new calls are allowed.
Ping A Ping is sent periodically for each gateway to the
corresponding server. The response received from the
server means that it is reachable. If no response is received
after the retransmission timer expires (configurable via the
Transmission Timeout field in “SIP Interop” on page 312),
the gateway considers the server as unreachable. In this
case, any call attempt through the gateway is refused. The
Pings are still sent when the server is unreachable and as
soon as it becomes reachable again, new calls are allowed.
2. Set the interval, in seconds, at which SIP Keep Alive requests using SIP OPTIONS or Ping are sent
to verify the server status in the Keep Alive Interval field.
3. Select the behaviour of the device when performing the keep alive action in the Keep Alive
Destination drop-down menu.
Table 106: SIP Keep Alive Destination Parameters
First SIP Destination Performs the keep alive action through the first SIP
destination. This corresponds to the outbound proxy host
when specified, otherwise it is the proxy host.
Alternate Destination Performs the keep alive action through the alternate
destination target (see “SIP Gateway Specific Keep Alive
Destinations” on page 164 for more details).
SIP Gateway Specific Keep Alive Destinations

This section allows you to override the default Keep Alive destination alternate target when the Keep Alive
Destination drop-down menu is set to Alternate Destination (see “Keep Alive” on page 163 for more details).
 To set specific keep alive destinations:

1. In the Keep Alive Destinations section of the Servers page, set the Alternate destination target
server FQDN and port for a specific SIP gateway in the default field.
192.168.0.5:5060

Outbound Proxy Loose Router Configuration Software Configuration Guide
Figure 68: SIP Servers – Specific Keep alive Targets

1
Outbound Proxy Loose Router Configuration
Standards Supported • RFC 3261: SIP: Session Initiation Protocol, section 6

• RFC 2543: SIP: Session Initiation Protocol
You can specify the type of routing of the outbound proxy configured in “SIP Servers Configuration” on
page 160.
 Default configurations that apply to all the endpoints of the Aastra unit.
You can define specific configurations for each endpoint in your Aastra unit. For instance, you could
enable a codec for all the endpoints of the Aastra unit and use the specific configuration parameters
to disable this same codec on one specific endpoint.
Using one or more specific parameter usually requires that you enable an override variable and set the specific
configuration you want to apply.
The following types are available:
Table 107: Outbound Proxy Router Status
Type Description
LooseRouter This is the most current method for SIP routing, as per RFC 3261, and will become the
standard behaviour once RFC 3261 compliance is achieved. See “Introduction” on
page 159 for details.
StrictRouter Pre-RFC 3261, RFC 2543 compatible SIP routing.
The initial route for all SIP requests contains the home domain proxy address (the Request-
URI). Requests are directed to the outbound proxy.
In other words, the Request-URI is constructed as usual, using the home domain proxy and
the user name, but is used in the route set. The Request-URI is filled with the outbound
proxy address.
Loose Router
A proxy is said to be loose routing if it follows the procedures defined in the RFC 3261 specification
(section 6) for processing of the Route header field. These procedures separate the destination of the
request (present in the Request-URI) from the set of proxies that need to be visited along the way
(present in the Route header field). A proxy compliant to these mechanisms is also known as a loose
router.

Chapter 22 - SIP Servers Outbound Proxy Loose Router Configuration
Table 107: Outbound Proxy Router Status (Continued)
Type Description
NoRouteHea Removes the route header from all SIP packets sent to an outbound proxy. This does not
der modify persistent TLS connection headers.
Note: The Router header will not be removed from the SIP packets if the unit is configured
to use the TLS Fallback feature. This feature requires the information of the SIP Outbound
Proxy in the SIP packet to work correctly.
 To set the outbound proxy router status:

1. In the sipEpMIB, set the defaultProxyOutboundType variable to the proper value.
sipEp.defaultProxyOutboundType="Value"
Table 108: Outbound Proxy Router Values
Value Meaning
100 LooseRouter
200 StrictRouter
300 NoRouteHeader
2. If you want to set a different routing type for one or more SIP gateways, set the following variables:
• gwSpecificproxyEnableConfig variable for the specific SIP gateway you want to
configure to enable.
• gwSpecificProxyOutboundType variable for the specific SIP gateway you want to
configure to the proper value.
You can also use the following lines in the CLI or a configuration script:
sipEp.gwSpecificProxy.EnableConfig[GatewayName="default"]="1"
sipEp.gwSpecificProxy.OutboundType[GatewayName="Specific_Gateway"]="Value"
where:
• Specific_Gateway is the name of the SIP gateway you want to configure.
• Value is the refresh router status as defined in Step 1.

C H A P T E R
23 SIP Registration
This chapter describes how to configure the registration parameters of the Aastra unit.

• RFC 3261: The Session Initiation Protocol (SIP)
• RFC 3863: Presence Information Data Format (PIDF)
• RFC 3903: Session Initiation Protocol (SIP) Extension for
Event State Publication
Endpoints Registration
Each endpoint of the Aastra unit has its own registration information. You can set information for each endpoint
such as its telephone number and friendly name.
Adding an endpoint registration triggers a warning message if the total number of registrations configured
reached the defined limit. See “Number of Registrations” on page 171 for more details.
 To set endpoints registration information:

1. In the web interface, click the SIP link, then the Registrations sub-link.
Figure 69: SIP – Registrations Web Page
2 3 4 5 6
2. In the Endpoints Registration and Subscription section of the Registrations page, enter a user name
for each endpoint in the User Name column.
The user name (such as a telephone number) uniquely identifies this endpoint in the domain. It is
used to create the Contact and From headers. The From header carries the permanent location (IP
address, home domain) where the endpoint is located. The Contact header carries the current
location (IP address) where the endpoint can be reached.
Contacts are registered to the registrar. This enables callers to be redirected to the endpoint’s
current location.
3. Enter another name for each endpoint in the Friendly Name column.

Chapter 23 - SIP Registration Endpoints Registration
This is a friendly name for the endpoint. It contains a descriptive version of the URI and is intended
to be displayed to a user interface.
4. Define whether or not the endpoint registration needs to register to the registrar in the Register
column.
An endpoint configured to register (set to Enable) will become unavailable for calls from or to SIP
when not registered.
You can define the behaviour of an endpoint when it becomes unavailable in the
defaultRegistrationUnregisteredBehavior MIB variable.
5. Define whether or not the endpoint needs to subscribe to a messaging system in the Messaging
drop-down menu.
The current state of the subscription is displayed in the Endpoints Messaging Subscription Status
table.
Table 109: MWI Subscription State
State Description
Unsubscribed The unit/endpoint is not subscribed and never tries to subscribe. This case
occurs if the network interface used by the SIP gateway is not up or the unit/
endpoint is locked.
Subscribing The subscription is currently trying to subscribe.
Subscribed The subscription is successfully subscribed.
Refreshing The subscription is trying to refresh.
Unreachable The last subscription attempt failed because the messaging server is
unreachable.
AuthFailed The last subscription attempt failed because authentication was not
successful.
Rejected The last subscription attempt failed because the messaging server rejects
the subscription.
ConfigError The last subscription attempt failed because it was badly configured. Check
if the username and the messaging host are not empty.
InvalidResponse The received 200 OK response contact does not match the contact of the
messaging server, or the 200 OK response for an unsubscribe contains a
contact.
You can enter the address of the Messaging server in “SIP Servers Configuration” on page 282.
6. Select on which SIP gateway the user configuration is applied in the Gateway Name drop-down
menu.
You must have SIP gateways already defined. See “Chapter 24 - SIP Gateways” on page 277 for
more details. If you select all, the configuration applies to all gateways available.
• To save your settings and refresh the registration now, click Submit & Refresh.
Contact Domain

Unit Registration Software Configuration Guide
You can set the host part of the SIP contact field. If an empty string is specified, the listening IP address is
used.
 To set the contact domain:

1. In the sipEpMIB, define the host part of the SIP contact field in the userAgentDomain variable.
sipEp.userAgentDomain=[value]
Unit Registration
Unit registration is used to register a contact not directly related to endpoints. This is generally used to indicate
to a registrar the IP location of the Aastra unit when it is used as a gateway.
Adding a unit registration triggers a warning message if the total number of registrations configured reached
the defined limit. See “Number of Registrations” on page 171 for more details.
 To set unit registration information:

1. In the Unit Registration section of the Registrations page, enter a user name in the User Name
column.
Figure 70: SIP Registrations – Unit Registration Section

1 2
The user name (such as a telephone number) uniquely identifies this user in the domain.
You can add a new user by clicking the button.
You can delete an existing user by clicking the button.
2. Select on which SIP gateway the user configuration is applied in the Gateway Name drop-down
menu.
You must have SIP gateways already defined. See “Chapter 24 - SIP Gateways” on page 277 for
more details. If you select all, the configuration applies to all gateways available.
Registration Configuration
This section allows you to define registration refresh parameters.

See “Additional Registration Refresh Parameters” on page 172 for more registration parameters.

Chapter 23 - SIP Registration Registration Configuration
 To set the registration configuration:

1. In the Registration Configuration section of the Registrations page, set the Default Registration
Refresh Time field with the time, in seconds, at which a registered unit begins updating its
registration before the registration expiration.
Figure 71: SIP Registrations – Registration Configuration Section
1
2
3
In SIP, a registration is valid for a period of time defined by the registrar. Once a unit is registered,
the SIP protocol requires the User Agent to refresh this registration before the registration expires.
Typically, this re-registration must be completed before the ongoing registration expires, so that the
User Agent's registration state does not change (i.e., remains 'registered').
For instance, if the parameter is set to 43 and the registration lasts one hour, the unit will send new
REGISTER requests 59 minutes and 17 seconds after receiving the registration acknowledgement
(43 seconds before the unit becomes unregistered).
Note: Normally, the Aastra unit cannot make or receive calls until the REGISTER has completed
successfully. Because the timeout for a SIP transaction in UDP is 32 seconds, it is possible to have an
ongoing re-REGISTER transaction at the same moment that the registration itself expires. This could
happen if the Default Registration Refresh Time field is set to a value lower than 32.
In that case, the user agent becomes unregistered, and will become registered again only when the re-
REGISTER request is answered with a positive response from the server. See “Gateway Specific
Registration Retry Time” on page 174 for a workaround if the unit cannot make calls during that period.
Setting this parameter to 0 means that the User Agent will fall into the 'unregistered' state BEFORE
sending the re-REGISTER requests.
This value MUST be lower than the value of the "expires" of the contact in the 200 OK response to
the REGISTER, otherwise the unit rapidly sends REGISTER requests continuously.
You can also set a different registration refresh time for one or more SIP gateways by using the MIB
parameters of the Aastra unit. See “Registration Refresh” on page 172 for more details.
2. Set the Proposed Expiration Value In Registration field with the suggested expiration delay, in
seconds, of a contact in the REGISTER request.
The SIP protocol allows an entity to specify the “expires” parameter of a contact in a REGISTER
request. The server can return this “expires” parameter in the 200 OK response or select another
“expires”. In the REGISTER request, the “expires” is a suggestion the entity makes.
The “expires” parameter indicates how long, in seconds, the user agent would like the binding to be
valid.
Available values are from 1 s to 86,400 s (one day).
This value does not modify the delay before a re-REGISTER.
• The delay is the “expires” of the contact in the 200 OK response to the REGISTER
request minus the value set in the Default Registration Refresh Time field.
• If the “expires” of the contact in the 200 OK response to the REGISTER is not present
or not properly formatted, then the delay is the default registration proposed expiration
value minus the value set in the Default Registration Refresh Time field.
Setting the parameter to 0 disables the expiration suggestion.
You can also set a different expiration delay for one or more SIP gateways by using the MIB
parameters of the Aastra unit. See “Registration Expiration” on page 173 for more details.
3. Set the Default Expiration Value in Registration field with the default registration expiration, in
seconds.
This value is used when the contact in a registration response contains no “expires” or the “expires”
is badly formatted. In this case, the delay before a re-REGISTER is the value set in this field minus
the value set in the Default Registration Refresh Time field (Step 1).

Number of Registrations Software Configuration Guide
You can also set a different expiration value in registration for one or more SIP gateways by using
the MIB parameters of the Aastra unit. See “Expiration Value in Registration” on page 173 for more
details.
Number of Registrations
The Aastra unit limits the total number of registrations to 100. The total number of registrations is the sum of
all the endpoints and gateways (“SIP Gateways Configuration” on page 277) pairs. The Aastra unit supports
a maximum of 5 gateways. An endpoint configured with "All" gateways generates as many pairs as the number
of gateways. In a setup with 3 gateways, one endpoint configured with "All" as the gateway name counts for
3 in the total number of registrations.
The registrations are enabled gateway by gateway until the limit is reached. Endpoints Registrations are used
first, then Unit Registrations. The remaining registrations are not registered and do not appear in the status
table. If you click the Submit And Refresh button and the configured number of registrations exceeds the
defined limit, a warning is displayed on the web interface (as well as in the CLI and SNMP interfaces) and a
syslog notify (Level Error) is sent.
Adding a gateway or an endpoint triggers a warning message if the total number of registrations configured
reached the defined limit.
Let’s suppose for instance that we have the current SIP Gateways configuration and the following SIP
Registration configuration:
Figure 72: Example, Gateway Configuration
Figure 73: Example, Registrations Configuration
The following table describes how to compute the total number of registrations for this example:
Table 110: Number of Registrations Example
Parameter Setting Nb of Registrations
Endpoint Registration 1 in Figure 73 Gateway Name set to alla 3

Endpoint Registration 2 in Figure 73 Gateway Name set to gw2 1
Unit Registration 1 in Figure 73 Gateway Name set to all 3
Unit Registration 2 in Figure 73 Gateway Name set to all 3
Unit Registration 3 in Figure 73 Gateway Name set to gw1 1
Unit Registration 4 in Figure 73 Gateway Name set to default 1

Chapter 23 - SIP Registration Additional Registration Refresh Parameters
Table 110: Number of Registrations Example (Continued)
Parameter Setting Nb of Registrations
Total Number of registrations 12

a. When the Gateway Name is set to all, this must be multiplied by the number of gateways set in Figure 72.
In this example, there are 3 gateways set.
Additional Registration Refresh Parameters
Default Registration Retry Time

You can configure the interval in seconds (s) on which a failed registration is retried.
This variable defines the time, relative to the failure of the registration, at which the device retries the
registration.
 To specify the default registration retry time value:

1. In the sipEpMIB, locate the registrationGroup folder.
2. Set the DefaultRegistrationRetryTime variable with the desired interval value.
sipEp.DefaultRegistrationRetryTime="Value"
where Value may be between 1 and 86400 seconds.

Registration Refresh
You can set the default registration refresh time in the web page (“Registration Configuration” on page 169),
but you can also set a different registration refresh time for one or more SIP gateways.
 To set registration refresh parameters:

2. If you want to set a different registration refresh time for one or more SIP gateways, set the following
variables:
• gwSpecificRegistrationEnableConfig variable for the specific SIP gateway you
want to configure to enable.

Additional Registration Refresh Parameters Software Configuration Guide
• gwSpecificRegistrationRefreshTime variable for the specific SIP gateway you want

to configure to the proper value.
sipEp.gwSpecificRegistration.EnableConfig[GatewayName="Specific_Gateway"]="1"
sipEp.gwSpecificRegistration.RefreshTime[GatewayName="Specific_Gateway"]="Value"
where:
• Value is the refresh time value.
Registration Expiration
You can set the default registration proposed expiration value in the web page (“Registration Configuration”
on page 169), but you can also set a different registration refresh time for one or more SIP gateways.
 To configure the registration expiration:

2. If you want to set a different registration refresh time for one or more SIP gateways, set the following
variables:
• gwSpecificRegistrationProposedExpirationValue variable for the specific SIP
gateway you want to configure to the proper value.
sipEp.gwSpecificRegistration.ProposedExpirationValue[GatewayName="Specific_Gatew
ay"]="Value"
where:
• Value is the expiration delay value.
This value does not modify the time before a re-REGISTER.
• The delay is the “expires” of the contact in the 200 OK response to the REGISTER
request minus the value set in the gwSpecificRegistrationRefreshTime parameter.
• If the “expires” of the contact in the 200 OK response to the REGISTER is not present
or not properly formatted, then the delay is the default registration proposed expiration
value minus the value set in the gwSpecificRegistrationRefreshTime parameter.
Expiration Value in Registration

You can set the default expiration value in registration in the web page (“Registration Configuration” on
page 169), but you can also set a different expiration value in registration for one or more SIP gateways.
This value is used when the contact in a registration response contains no “expires” or the “expires” is badly
formatted. In this case, the delay before a re-REGISTER is the value set in this field minus the value set in the
in the 'RefreshTime' variable (“Registration Refresh” on page 172).
 To configure the expiration value in registration for a specific gateway:

2. To set a different expiration value in registration for one or more SIP gateways, set the following
variables:
• gwSpecificRegistrationExpirationValue variable for the specific SIP gateway you
want to configure to the proper value.

3. To set a different expiration value in registration for one or more SIP gateways, put the following
lines in the configuration script:
sipEp.gwSpecificRegistration.ExpirationValue[GatewayName="Specific_Gateway"]="Va
lue"
where:
• Value is the expiration value in registration value.
Gateway Specific Registration Retry Time

You can set a different Registration Retry Time for one or more SIP gateways.
This variable defines the time, relative to the failure of the registration, at which the SIP gateway retries the
registration.
 To specify the registration retry time value for a specific gateway:

2. To set a different registration retry time for one or more SIP gateways, set the following variables:
• gwSpecificRegistrationRetryTime variable for the specific SIP gateway you want to
3. To set a different expiration value in registration for one or more SIP gateways, put the following
lines in the configuration script:
sipEp.gwSpecificRegistrationRetryTime[GatewayName="Specific_Gateway"]="Value"
where:
• Value is the expiration value in registration retry time.
Unregistered Endpoint Behaviour

You can specify whether an endpoint should remain enabled or not when not registered. This is useful if you
want your users to be able to make calls even if the endpoint is not registered with a SIP server.
The following values are supported:
Table 111: Unregistered Endpoint Behaviour Parameters
Value Description
disablePort When the endpoint is not registered, it is disabled. The user cannot make or receive calls.
Picking up the handset yields a fast busy tone, and incoming INVITEs receive a “403
Forbidden” response.
enablePort When the endpoint is not registered, it is still enabled. The user can receive and initiate
outgoing calls. Note that because the endpoint is not registered with a registrar, its public
address is not available to the outside world; the endpoint will most likely be unreachable
except through direct IP calling.

Additional Registration Refresh Parameters Software Configuration Guide
 To specify unregistered endpoint behaviour:

2. Set the defaultRegistrationUnregisteredBehavior variable.
sipEp.defaultRegistrationUnregisteredBehavior="Value"
where Value may be as follows:.
Table 112: Unregistered Endpoint Behaviour Values
Value Meaning
0 disablePort
1 enablePort
3. If you want to set a different behaviour for one or more SIP gateways, set the following variables:
• gwSpecificRegistrationUnregisteredBehavior variable for the specific SIP
gateway you want to configure to the proper value.
sipEp.gwSpecificRegistration.UnregisteredBehavior[GatewayName="Specific_Gateway"
]="Value"
where:
• Value is one of the values described in Step 2.
Unregistered User Behaviour

You can specify whether the SIP gateway state should be affected or not by the unit registrations state.
Table 113: Unregistered User Behaviour Parameters
Value Description
NoEffect The unit registrations state has no effect on the SIP gateway state.
DisableGate The SIP gateway goes in the 'unregistered' state when all unit registrations are not in the
way 'registered' state. The 'unregistered' state indicates some registrations that are mandatory
for this gateway failed.
 To specify unregistered user behaviour:

2. Set the defaultUserRegistrationUnregisteredBehavior variable.
sipEp.defaultUserRegistrationUnregisteredBehavior="Value"
Table 114: Unregistered User Behaviour Values
Value Meaning
100 NoEffect

Table 114: Unregistered User Behaviour Values (Continued)
Value Meaning
200 DisableGateway
Behaviour on Initial-Registration Reception

You can configure the behaviour of the Aastra unit upon reception of a 380 or 504 carrying an XML body with
a specified 'initial-registration' action.
Table 115: Behaviour on Initial-Registration Reception Parameters
Value Description
NoRegistration No registration refresh are sent upon reception of the message.

EndpointRegistration Registration refresh of the endpoint associated with the call is sent upon
reception of the message.
UnitRegistration Registration refresh of all the usernames configured as 'unit registration'
are sent upon reception of the message.
UnitAndEndpointRegistration Registration refresh of the endpoint associated with the call and of all the
usernames configured as 'unit registration' are sent upon reception of the
message.
 To specify the behaviour on Initial-Registration reception:

2. Set the behaviorOnInitialRegistrationReception variable with the proper behaviour.
sipEp.behaviorOnInitialRegistrationReception="Value"
Table 116: Behaviour on Initial-Registration Reception Values
Value Meaning
100 NoRegistration
200 EndpointRegistration
300 UnitRegistration
400 UnitAndEndpointRegistration
If the registration(s) succeed, then the call is re-attempted.

If the registration(s) fail, then the call is terminated.
3. Set the registrationDelayOnInitialRegistrationReception variable with the registration
delay, in milliseconds, on Initial-Registration Reception.
This variable configures the time interval between the unregistration confirmation (or final response)
and the registration attempt that follows.
This variable is only used when behaviorOnInitialRegistrationReception is configured to a
value other than 'NoRegistration'.
Note: This variable only applies on registration refresh triggered by the

behaviorOnInitialRegistrationReception feature.

SIP User Agent Header Software Configuration Guide
sipEp.registrationDelayOnInitialRegistrationReception="Value"
Registration Delay Value

The quality of calls may be altered if a large quantity of registrations, more than 100, is requested at the same
time. To avoid this situation, you can configure the maximum number of seconds that the system uses to apply
a random algorithm, which is used to determine a delay before requesting a user registration or an endpoint
registration.
When the value is 0, the request registration is done immediately.
Note: The random algorithm applies individually to all registrations, meaning registrations order may not
follow their corresponding index.
 To specify the registration delay value:

1. In the sipEpMIB, set the interopRegistrationDelayValue variable with the proper delay value.
sipEp.interopRegistrationDelayValue="Value"
where Value may be between 0 and 600 seconds.
SIP User Agent Header
The User-Agent header field contains information about the user agent client originating the request. For
instance, the information of the User-Agent header could be something like the following:
User-Agent: Softphone Beta1.5
You can specify whether or not the Aastra unit sends this information when establishing a communication.
 To enable sending the SIP User Agent header:

1. In the sipEpMIB, set the interopSendUAHeaderEnable variable to enable.
sipEp.interopSendUaHeaderEnable="1"

Chapter 23 - SIP Registration SIP User Agent Header

C H A P T E R
24 SIP Authentication
This chapter describes how to configure authentication parameters of the Aastra unit.
Standards Supported • Basic and Digest authentication as per RFC 3261
Caution: The SIP > Authentication page is not accessible if you have the User or Observer access right.
See “Users” on page 591 for more details.
Authentication Configuration
Authentication information allows you to add some level of security to the Aastra unit endpoints by setting user
names and passwords.
You can add four types of authentication information:
Table 117: Authentication Information
Authentication Description
endpoint-specific Applies only to challenges received for SIP requests related to a specific endpoint. For
instance, the registration associated with the endpoint in the user agent table or the
INVITE sent to initiate a call from the endpoint. You can define several user names and
passwords for each endpoint of the Aastra unit. An endpoint can thus register with
several different realms.
gateway-specific Applies only to challenges received for SIP requests on a specific SIP gateway. You can
define several user names and passwords for each endpoint of the Aastra unit. An
endpoint can thus register with several different realms.
unit Applies to all challenges received for SIP dialog. You can define several user names
and passwords for the Aastra unit. These user names and passwords apply to all
endpoints of the unit.
user name- Applies only to challenges for a context that uses a specific user name.
specific
The Authentication table may have between 20 and 100 rows. Each of these rows can either be associated
with the unit, a specific gateway, a specific endpoint, or a specific user name. If you have less than 20 rows,
the Aastra unit automatically adds new rows up to the minimum of 20.
When a challenge occurs (either 401 or 407), the first entry in the Authentication table that matches the user
name/password request is used to reply to the challenge. You can configure the use name and password in
the web interface. The order of the tried entries in the Authentication table is from the first row to the last row.
The challenge matches an authentication entry if the realm of the challenge matches the realm specified in
the Realm field or if the Validate Realm field is set to disable. For each entry matching certain criteria
(described below), the challenge is replied with the entry's user name and password. If no entry matches the
criteria, the authentication fails. To match the authentication request, the entry must also meet one of the
following criteria:
 The challenge needs to be for a SIP request related to the endpoint specified in the Endpoint
column if the corresponding Apply To column is set to Endpoint.
 The challenge needs to be for a SIP request performed on the SIP gateway specified in the
Gateway column if the corresponding Apply To column is set to Gateway.

Chapter 24 - SIP Authentication Authentication Configuration
 The challenge needs to be for a context that uses the user name specified in the User Name
field if the corresponding Apply To column is set to Usename. The user name associated with
a context is:
• the user name of the FROM if the context sent the original SIP request, or
• the user name of the request URI if the context received the original SIP request
 The challenge applies to a unit if the corresponding Apply To column is set to Unit.
Creating/Editing an Authentication Entry

The web interface allows you to create authentication entries or modify the parameters of an existing one.
 To create or edit SIP authentication parameters:

1. In the web interface, click the SIP link, then the Authentication sub-link.
Figure 74: SIP Configuration – Authentication Web Page
2. Do one of the following:

• If you want to add an authentication entry before an existing entry, locate the proper
row in the table and click the button of this row.
• If you want to add an authentication entry at the end of the existing rows, click the
button at the bottom right of the Authentication section.
• If you want to add several authentication entries at the same time, enter the number of
entries you want to add in the Number of rows to add at the bottom of the page.
• If you want to edit a single authentication entry, locate the proper row in the table and
click the button.
• If you want to edit a several authentication entries of the current page at the same time,
click the Edit All Entries button at the bottom of the page.
This brings you to the proper Authentication panel.

Authentication Configuration Software Configuration Guide
Table 118: Authentication Panel – Single Entry
3 4 5 6 7 8 9
Table 119: Authentication Panel – Page
3 4 5 6 7 8 9
3. Select which criterion to use for matching an authentication request with an authentication entry in
the Apply to column.
Table 120: Authentication Entity
Unit The authentication entry is used on all challenges.

Endpoint The authentication entry used for all challenges related to
a specific endpoint.
Gateway The authentication entry is used for all challenges related
to a specific SIP gateway.
Username The authentication entry is used for all challenges related
to a specific user name. Only the username part is used if
the value has the format 'username@domain'.
4. Enter a string that identifies an endpoint in other tables in the Endpoint column.
This field is available only if you have selected the Endpoint entity in the previous step for the
specific row.
5. Enter a string that identifies a SIP gateway in other tables in the Gateway column.
This field is available only if you have selected the Gateway entity in the Apply to column for the
specific row.

Chapter 24 - SIP Authentication Authentication Configuration
6. Select whether or not the current credentials are valid for any realm in the corresponding Validate
Realm drop-down menu.
Table 121: Realm Authentication Parameters
Disable The current credentials are valid for any realm. The corresponding Realm field is
read-only and cannot be modified.
Enable The credentials are used only for a specific realm set in the corresponding Realm
field.
7. Enter a realm for each authentication row in the Realm column.

When authentication information is required from users, the realm identifies who requested it.
8. Enter a string that uniquely identifies this endpoint in the realm in the User Name column.
9. Enter a user password in the Password column.
Registration.
Moving an Authentication Entry

The order of the tried entries in the Authentication table is from the first row to the last row. The rows sequence
is thus very important. If you want the unit to try to match one row before another one, you must put that row
first.
 To move an authentication entry up or down:

1. Either click the or arrow of the row you want to move until the entry is properly located.
Deleting an Authentication Entry

You can delete an authentication row from the table in the web interface.
 To delete an authentication entry:


C H A P T E R
25 SIP Transport Parameters
This chapter describes the SIP transport parameters you can set.
SIP Transport Type
Standards Supported • RFC 2246: The TLS Protocol Version 1.0

• RFC 3261: SIP, Session Initiation Protocol
You can globally set the transport type for all the endpoints of the Aastra unit to either UDP (User Datagram
Protocol), TCP (Transmission Control Protocol), or TLS (Transport Layer Security).
The Aastra unit will include its supported transports in its registrations.
Please note that RFC 3261 states the implementations must be able to handle messages up to the maximum
datagram packet size. For UDP, this size is 65,535 bytes, including IP and UDP headers. However, the
maximum datagram packet size the Aastra unit supports for a SIP request or response is 5120 bytes excluding
the IP and UDP headers. This should be enough, as a packet is rarely bigger than 2500 bytes.
 To set the SIP transport type parameters:

1. In the web interface, click the SIP link, then the Transport sub-link.
Figure 75: SIP Configuration – Transport Web Page
2
3
4
5
6
7 8 7 8 7 8
2. In the General Configuration section, enable or disable the transport registration in the Add SIP
Transport in Registration drop-down menu.
When enabled, the Aastra unit includes its supported transports in its registrations. It registers with
one contact for each transport that is currently enabled. Each of these contacts contains a
“transport” parameter.

Chapter 25 - SIP Transport Parameters SIP Transport Type
This is especially useful for a system where there are no SRV records configured to use a
predefined transport order for receiving requests. When sending a request, the unit either follows
the SRV configuration, or, if not available, any transport parameter received from a redirection or
from a configured SIP URL.
Note: If the Aastra unit has the following configuration:

• the Add SIP Transport in Registration drop-down menu is set to Disable
• the UDP transport type is disabled
• the TCP transport type is enabled
The unit will not work properly unless the SIP server uses the TCP transport type by default.
This is also true if the Aastra unit has the TCP transport disabled and the UDP transport enabled. In this
case, the unit will not work properly unless the SIP server uses the UDP transport protocol by default.
3. Indicate whether or not the unit must include its supported transport in the Contact header in the
Add SIP Transport in Contact Header drop-down menu.
The supported transports are included in all SIP messages that have the Contact header, except
for the REGISTER message.
Available values are Enable and Disable. If you set the menu to Enable, the Aastra unit will send
SIP messages with the “transport” parameter in the Contact header set to:
• transport=tcp when TCP is enabled and UDP is disabled
• transport=udp when UDP is enabled and TCP disabled
• no transport parameter when both TCP and UDP are enabled
• transport=tls when secure transport (TLS) is selected
4. Define the base port used to establish TLS persistent connections with SIP servers when the TLS
transport is enabled in the Persistent TLS Base Port field.
5. Set the time interval, in seconds, before retrying the establishment of a TLS persistent connection
in the Persistent TLS Retry Interval field.
This is the interval that the Aastra unit waits before retrying periodically to establish a TLS persistent
connection using a single IP address or a FQDN. This timer is started when a TLS persistent
connection goes down or fails to connect to the destination. The TLS persistent connect timeout
applies only to TLS persistent connections.
When the destination is a single IP address and the TLS persistent connection goes down or fails
to establish, the timer is started. When the timer expires, the Aastra unit attempts to re-establish the
TLS persistent connection.
When the destination is a FQDN and the TLS persistent connection goes down or fails to establish
with the higher priority target received from a DNS answer, the timer is started and the lower priority
targets are attempted. When the timer expires, a new DNS request is sent and depending on the
DNS answer, the Aastra unit retries to establish the TLS persistent connection with the higher
priority target. The timer is unique for all TLS persistent connections using the same FQDN. This
means that the timer is not restarted when a connection using a lower priority target fails while a
connection using a higher priority target has already failed.
6. In the TLS Trusted Certificate Level field, define how a peer certificate is considered trusted for a
TLS connection.
Table 122: Certificate Trust Level for TLS Connections Parameters
Locally A certificate is considered trusted when the certificate authority (CA) that signed the
Trusted peer certificate is present in the Others Certificates table (see “Chapter 46 -
Certificates Management” on page 557 for more details). The certificate revocation
status is not verified.

Additional Transport Parameters Software Configuration Guide
Table 122: Certificate Trust Level for TLS Connections Parameters (Continued)
OCSP A certificate is considered trusted when it is locally trusted and is not revoked by its
Optional certificate authority (CA). The certificate revocation status is queried using the
Online Certificate Status Protocol (OCSP). If the OCSP server is not available or
the verification status is unknown, the certificate is considered trusted.
OCSP A certificate is considered trusted when it is locally trusted and is not revoked by its
Mandatory certificate authority (CA). The certificate revocation status is queried using the
Online Certificate Status Protocol (OCSP). If the OCSP server is not available or
the verification status is unknown, the certificate is considered not trusted.
7. Set the TCP Connect Timeout field with the maximum time, in seconds, the unit should try to
establish a TCP connection to SIP hosts.
This timeout value is useful to have a faster detection of unreachable remote hosts. This timer can
also affect the TLS connection establishment time.
8. In the Protocol Configuration section, enable or disable the UDP, TCP, and TLS transport type to
use in their corresponding drop-down menu.
UDP and TCP are mutually exclusive with TLS. Activating TLS automatically disables these
unsecure protocols.
The successful configuration of a secure transport requires a little more than the activation of the
TLS protocol itself. You need to:
• synchronize the time in the unit (see “Time Configuration” on page 94 & “SNTP
Configuration” on page 93 for more details).
• install the security certificates used to authenticate the server to which you will connect
(see “Chapter 46 - Certificates Management” on page 557 for more details).
• Use secure media (see “Security” on page 201 for more details).
• configure the unit so that a “transport=tls” parameter is added to the Contact header of
your SIP requests (see Step 3).
Caution: If you have enabled Secure RTP (SRTP) on at least one line, it is acceptable to have the secure
SIP transport (TLS) disabled for testing purposes. However, you must never use this configuration in a
production environment, since an attacker could easily break it. Enabling TLS for SIP Transport is strongly
recommended and is usually mandatory for security interoperability with third-party equipment.
9. Set the priority order of each transport type in the corresponding QValue field.
A qvalue parameter is added to each contact. The qvalue gives each transport a weight, indicating
the degree of preference for that transport. A higher value means higher preference.
The format of the qvalue string must follow the RFC 3261 ABNF (a floating point value between
0.000 and 1.000). If you specify an empty string, no qvalue is set in the contacts.
Additional Transport Parameters

Chapter 25 - SIP Transport Parameters Additional Transport Parameters
UDP Source Port Behaviour

You can configure whether or not the Aastra unit always uses the same local port (the port on which it is
listening for incoming packets) when sending SIP traffic over UDP. This is called symmetric UDP source port.
Symmetric UDP ports are sometimes needed to traverse NAT/Firewall devices.
When changing this setting, all destinations are automatically sent out of the penalty box, when applicable.
Table 123: UDP Source Port Parameters
disable The SIP signalling over UDP uses a randomly-generated originating port. ICMP errors are
processed correctly.
enable The SIP signalling sent over UDP originates from the same port as the port on which the user
agent is listening. ICMP messages are not processed, which means that unreachable targets
will take longer to detect.
 To set the UDP source port behaviour:

1. In the sipEpMIB, set whether or not the unit uses the symmetric source port feature in the
interopSymmetricUdpSourcePortEnable variable.
sipEp.interopSymmetricUdpSourcePortEnable="Value"
Table 124: UDP Source Port Values
Value Meaning
0 disable
1 enable
2. Restart the SipEp service by accessing the scmMIB and setting the serviceCommandsRestart
variable for the SipEp service to restart.
scm.serviceCommands.Restart[Name=SipEp]="10"
TLS Client Authentication

When acting as a TLS server, it is customary not to request from the clients that they authenticate themselves
via the TLS protocol. However, if mutual authentication is required between client and server, you can set the
Aastra unit so that it requests client authentication when acting as a TLS server.
Table 125: TLS Client Authentication Parameters
disable The Aastra unit does not require TLS clients to provide their host certificate for the connection
to be allowed. This is the default value.
enable The TLS clients must provide their host certificate for the connection to be allowed. In this case,
the level of security used to validate the host certificate is TrustedCertificate, whatever the
value set in the Certificate Validation drop-down menu of the TLS Interop section ( SIP >
Interop web page). See “TLS Interop” on page 318 for more details.

Additional Transport Parameters Software Configuration Guide
 To set TLS client authentication:

1. In the sipEpMIB, set whether or not the Aastra unit requests client authentication when acting as a
TLS server in the interopTlsClientAuthenticationEnable variable.
sipEp.interopTlsClientAuthenticationEnable="Value"
Table 126: TLS Client Authentication Values
Value Meaning
0 disable
1 enable
Force DNS NAPTR In TLS

The Aastra unit allows you to force a DNS NAPTR request when the SIP transport is TLS.
This variable only applies to calls over TLS when the Supported DNS Queries drop-down menu of the SIP >
Misc page is set to NAPTR (see “DNS Configuration” on page 348 for more details).
Table 127: Force DNS NAPTR in TLS Parameters
disable The DNS SRV request is sent directly with the SIP transport in SIP URI as recommended in
RFC 3263, section 4.1.
enable A DNS NAPTR request is sent to obtain the DNS record associated with SIP over TLS. An SRV
request is performed afterward. If no SIP over TLS entry is returned, the call fails.
 To force DNS NAPTR in TLS:

1. In the sipEpMIB, set whether or not to force a DNS NAPTR request in the
InteropForceDnsNaptrInTls variable.
sipEp.interopForceDnsNaptrInTls="Value"
Table 128: Force DNS NAPTR in TLS Values
Value Meaning
0 disable
1 enable

Chapter 25 - SIP Transport Parameters Additional Transport Parameters

C H A P T E R
26 Interop Parameters
This chapter describes the interop parameters that allow the Aastra unit to properly work, communicate, or
connect with specific IP devices.
Standards Supported • draft-ietf-sipping-realtimefax-00

• ITU-T Recommendation T.38, section D.2.3
• RFC 3264: An Offer/Answer Model with the Session
Description Protocol (SDP)
• RFC 3515: The Session Initiation Protocol (SIP) Refer
Method
Behavior on T.38 INVITE Not Accepted
This section describes the unit’s behaviour after receiving an error to a SIP INVITE for T.38 fax.
 To set the T.38 interop parameters:

1. In the web interface, click the SIP link, then the Interop sub-link.
Figure 76: SIP – Interop Web Page
2. In the Behavior on T.38 INVITE Not Accepted section, for each of 406, 415, 488, and 606 SIP code,
set the behaviour after receiving the code in the error response to an INVITE for T.38 fax in the
corresponding Behavior drop-down menu.
Table 129: Behavior on T.38 INVITE Not Accepted Parameters
Behavior Description
Drop Call The call is dropped by sending a BYE.

ReInviteForClearChannelOnly A re-INVITE is sent with audio codecs that support clear channel
faxes.
Re-Establish Audio A re-INVITE is sent to re-establish the audio path. Also, fax
detection is disabled for the remainder of the call.
UsePreviousMediaNegotiation No re-INVITE is sent and the audio codec from the last
successful negotiation is used. For the remainder of the call,
T.38 is disabled and fax detection may trigger a switch to a clear
channel codec that was available in the last successful
negotiation.

Chapter 26 - Interop Parameters SIP Interop
SIP Interop
This section describes the SIP interop parameters of the Aastra unit
 To set the SIP interop parameters:

1. In the SIP Interop section of the Interop page, set whether or not the “x-Siemens-Call-Type” header
is added to the SIP packets sent by the unit in the Secure Header drop-down header.
You can set the Aastra unit so that it triggers the addition of the “x-Siemens-Call-Type” header to
the SIP packets sent by the unit when secure transport is in use.
Table 130: Secure Transport Header Parameters
disable The “x-Siemens-Call-Type” header is not added to the SIP packets sent by the
unit.
enable The “x-Siemens-Call-Type” header is added to the SIP packets sent by the unit, and
assigned the value “ST-secure”, as soon as secure transport and secure payload
are being used. If secure transport or secure payload are not used, the header is
not added.
Figure 77: SIP Interop Section
1
2
3
4
5
6
7
8
2. Select the username to use when the username is empty or undefined in the Default Username
Value drop-down menu.
Table 131: Default Username Value
Anonymous Sets the username to “anonymous”.

Host Sets the username to the same value as the host.
3. Define the behaviour of the Aastra unit when answering a SIP OPTIONS request in the OPTIONS
Method Support drop-down menu.
Table 132: OPTIONS Method Support Parameters
None The Aastra unit responds with an error 405 Method not
allowed.
AlwaysOK The Aastra unit responds with a 200 OK regardless of the
content of the OPTIONS request.

SIP Interop Software Configuration Guide
4. Define whether or not the SIP OPTIONS requests should be ignored when all endpoints are
unusable in the Ignore OPTONS on no usable endpoints drop-down menu.
Table 133: Ignore SIP Options Parameters
Enable The unit ignores SIP OPTIONS requests when all

endpoints are unusable. When at least one endpoint is
usable, then the SIP OPTIONS requests are answered
as configured in the OPTIONS Method Support drop-
down menu (see Step 10).
Disable The SIP OPTIONS requests are answered as configured
in the OPTIONS Method Support drop-down menu (see
Step 10) regardless of the state of the endpoints.
Note that this feature may be influenced by whether or not you have enabled the Monitor Link State
parameter. For more information:
• ISDN PRI interface: “PRI Configuration” on page 184
• ISDN BRI interface: “BRI Configuration” on page 195
• R2 PRI interface: “R2 Channel Associated Signaling” on page 224
5. Set the value of the user parameter in SIP URIs sent by the unit in the SIP URI User Parameter
Value field.
If you leave the field empty, the parameter is not added.
E.g : sip:[email protected];user=InteropSipUriUserParameterValue
Note that when the Map Plus To TON International drop-down menu is set to Enable, the
parameter's value might be overwritten (“Misc Interop” on page 196).
6. Set the Behavior On Machine Detection drop-down menu with the SIP device’s behavior when a
machine (fax or modem) is detected during a call.
Table 134: Behavior on Machine Detection Parameters
Re-INVITE On Fax T38 Only A SIP re-INVITE is sent only on a fax detection and T.38
is enabled.
Re-INVITE On No Negotiated Data A SIP re-INVITE is sent on a fax or modem detection if no
Codec data codec was previously negotiated in the original SDP
negotiation. In the case where at least one data codec
was previously negotiated in the SDP negotiation, the
device switches silently to a data codec without sending a
SIP re-INVITE. Note sthat if there is no data codec
enabled on the device, no SIP re-INVITE is sent and the
call is dropped by sending a BYE.
Re-INVITE Unconditional A SIP re-INVITE is sent with data codecs upon detection
of a fax or modem even if a data codec was negotiated in
the initial offer-answer. The T.38 codec is offered if it is
enabled and a fax is detected.
See “Data Codec Selection Procedure” on page 221 for more details on the procedure the Aastra
unit follows when selecting data codec.

Chapter 26 - Interop Parameters SDP Interop
7. Set the Registration Contact Matching field with the matching behaviour for the contact header
received in positive responses to REGISTER requests sent by the unit.
Table 135: Registration Contact Matching Parameters
Strict Matches the complete contact's SIP URI including any URI parameters, if any, as
per RFC 3261 sections '10.2.4 Refreshing Bindings' and '19.1.4 URI Comparison'.
The contact's SIP URI of a 2XX positive response MUST match the contact's SIP
URI of the REGISTER request.
IgnoreUriP Matches the username and the host port part of the contact's SIP URI. All URI
arams parameters are ignored.
8. Set the Transmission Timeout field with the time to wait for a response or an ACK before
considering a transaction timed out.
This corresponds to timers B, F and H for all transport protocols and timer J for UDP. These timers
are defined in section A of RFC 3261.
This timeout affects the number of retransmissions. Retransmissions continue to follow the timing
guidelines described in RFC 3261.
If a DNS SRV answer contains more than one entry, the Aastra unit will try these entries if the entry
initially selected does not work. You can configure the maximum time, in seconds, to spend waiting
for answers to messages, from a single source. Retransmissions still follow the algorithm proposed
in RFC 3261, but the total wait time can be overridden by using this feature.
For example, if you are using DNS SRV and more than one entry are present, this timeout is the
time it takes before trying the second entry.
Available values are from 1 to 32 seconds.
SDP Interop
Standards Supported • RFC 3264: An Offer/Answer Model with Session Description

Protocol (SDP)
This section describes the SDP interop parameters of the Aastra unit.

SDP Interop Software Configuration Guide
 To set the SDP interop parameters:

1. In the SDP Interop section of the Interop page, Offer Answer Model part, select the codec
negotiation rule when generating a SDP answer in the Answer Codec Negotiation drop-down menu.
Table 136: Answer Codec Negotiation Parameters
All Common - Local When generating an answer to an offered session, all common codecs
Priority are listed in the local order of priority. The local priority is defined for each
codec in the Telephony > CODECS page – by clicking the button of
each codec and looking in the Voice Priority and Data Priority fields. See
“Chapter 14 - Voice & Fax Codecs Configuration” on page 181 for more
details.
First Common - When generating an answer to an offered session, only the first common
Local Priority codec with the higher local priority is listed. The local priority is defined
for each codec in the Telephony > CODECS page – by clicking the
button of each codec and looking in the Voice Priority and Data Priority
fields. See “Chapter 14 - Voice & Fax Codecs Configuration” on
All Common - Peer When generating an answer to an offered session, all common codecs
Priority are listed. The codecs order is the same as in the peer offer.
First Common - When generating an answer to an offered session, only the first common
Peer Priority codec is listed. The codecs order is the same as in the peer offer.
Figure 78: SDP Interop Section
1
2
3
4
5
6
7
8
2. Select whether or not the Aastra unit requires strict adherence to RFC 3264 when receiving an
answer from the peer when negotiating capabilities for the establishment of a media session in the
Enforce Offer Answer Model drop-down menu.
Table 137: Offer/Answer Model Parameters
Disable The peer can freely:

• Send back a brand new list of codecs or add new ones to the offered
list.
• Add new media lines.
As long as at least one codec sent back was present in the initial offer, the call
is allowed to go on. Any media line added by the peer is simply ignored.

Chapter 26 - Interop Parameters SDP Interop
Table 137: Offer/Answer Model Parameters
Enable The following guidelines from the Offer-Answer Model must be strictly
followed. An answer must:
• Include at least one codec from the list that the Aastra unit sent in
the offer.
• Contain the same number of media lines that the unit put in its offer.
Otherwise, the answer is rejected and the unit ends the call. This is the default
value.
3. Define the behaviour of the Aastra unit when receiving less media announcements in the response
than in the offer in the Allow Less Media In Response drop-down menu.
Table 138: Less Media Announcements Parameters
Disable The Aastra unit rejects the response with less media announcements than in
the offer.
Enable The Aastra unit tries to find matching media when the response contains less
media announcement than in the offer. This is a deviation from the Offer/
Answer model.
4. Define the behaviour of the Aastra unit when receiving a SDP answer activating a media that had
been previously deactivated in the offer in the Allow Media Reactivation in Answer drop-down
menu.
Table 139: Media Reactivation Parameters
Enable A media reactivated in an incoming answer is ignored. This behaviour goes

against the SDP Offer/Answer model described by IETF RFC 3264.
Disable A media reactivated in an incoming answer ends the current media negotiation
and the call. This behaviour follows the SDP Offer/Answer model described by
IETF RFC 3264.
5. In the Multiple Active Media part, define the behaviour of the Aastra unit when offering media or
answering to a media offer with audio and image negotiation in the Allow Audio and Image
Negotiation drop-down menu.
Table 140: Audio and Image Negotiation Parameters
Enable The unit offers audio and image media simultaneously in

outgoing SDP offers and transits to T.38 mode upon
reception of a T.38 packet. Also, when the unit answers
positively to a SDP offer with audio and image, it transits
to T.38 mode upon reception of a T.38 packet.
Disable Outgoing offers never include image and audio
simultaneously. Incoming offers with audio and image
media with a non-zero port are considered as offering
only audio.

TLS Interop Software Configuration Guide
6. Define the behaviour of the Aastra unit when answering a request offering more than one active
media in the Allow Multiple Active Media in Answer drop-down menu.
Figure 79: Allow Multiple Active Media in Answer
disable The answer contains only one active media. The media specified as active
in the answer is the top-most matching one in the offer. Other media are
set to inactive.
enable Each matching active media in the offer is specified as active in the
answer. Other media are set to inactive
7. In the Other part, define how to set the direction attribute and the connection address in the SDP
when answering a hold offer with the direction attribute “sendonly” in the On Hold SDP Stream
Direction in Answer drop-down menu.
The following parameters are supported:
Table 141: “sendonly” Direction Attribute
inactive The stream is marked as inactive and if the stream uses IPv4, the
connection address is set to '0.0.0.0'.
revconly If the stream is currently active or receive only, it is marked as recvonly
and the connection address is set to the IP address of the unit.
If the stream is currently send only or inactive, it is marked as inactive
and if the stream uses IPv4, the connection address is set to '0.0.0.0'.
This method is in conformance with RFC 3264.
In both cases, no direction attribute is present in the SDP if the

interopSdpDirectionAttributeEnable variable is set to disable (see “Direction Attribute” on
Note: If you are experiencing media negotiation problems (because the Aastra unit sends a BYE after
receiving a 200 OK), try to set the Enforce Offer Answer Model value to Disable and the Allow Less Media
In Response value to Enable.
TLS Interop
This section describes the TLS interop parameters of the Aastra unit.
 To set the TLS interop parameters:

1. In the TLS Interop section of the Interop page, select the level of security used to validate the TLS
server certificate when the unit is acting as a TLS client in the Certificate Validation drop-down
menu.
Figure 80: TLS Interop Section
Note: This parameter has no effect on the TLS client authentication when the unit is acting as a TLS server
(see the interopTlsClientAuthenticationEnable variable in “TLS Client Authentication” on page 308).

Chapter 26 - Interop Parameters Misc Interop

Table 142: TLS Certificate Validation Parameters
No Validation No validation of the peer certificate is performed. All TLS connections are
accepted without any verification. Note that at least one certificate must be
returned by the peer even if no validation is made. This option provides no
security and should be restricted to a lab use only.
Trusted Allows a TLS connection only if the peer certificate is trusted. A certificate is
Certificate considered trusted when the certificate authority (CA) that signed the peer
certificate is present in the Management > Certificates page (“Chapter 46 -
Certificates Management” on page 557). This option provides a minimum level
of security and should be restricted to a lab use only.
Dns Srv Allows a TLS connection if the peer certificate is trusted and contains a known
Response host name. A known host name can be the FQDN or IP address configured as
the SIP server, or can also be returned by a DNS SRV request. In this case,
the match is performed against the DNS response name. If it matches either
one of the Subject Alternate Name (SAN) or Common Name (CN) in the peer
certificate, the connection is allowed. This option provides an acceptable level
of security, but not as good as Host Name.
HostName Allows a TLS connection if the peer certificate is trusted and contains a known
host name. A known host name can only be the FQDN or IP address
configured as the SIP server. If it matches either one of the Subject Alternate
Name (SAN) or Common Name (CN) in the peer certificate, the connection is
allowed. This option provides the highest level of security.
Misc Interop
This section describes miscellaneous interop parameters of the Aastra unit.
 To set the Misc interop parameters:

1. In the Misc Interop section of the Interop page, select whether or not the Aastra unit enables the
mapping between the “+” prefix of the user name and the “type of number” property in the Map Plus
To TON International drop-down menu.
When enabled, the service has the following behaviour:
• For a call to SIP, the Aastra unit prefixes the user name with '+' if the call has the call
property “type of number” set to international. The unit also adds the “user” parameter
with the value “phone” to the SIP URI. For instance:
sip:[email protected];user=phone.
• For a call from SIP, the Aastra unit sets the call property “type of number” to
international if the user name has the prefix '+'.
Figure 81: Misc Interop Section
1
2
3
4

Additional Interop Parameters Software Configuration Guide
2. Define the Ignore Plus in Username drop-down menu as to whether or not the plus (+) character is
ignored when attempting to match a challenge username with usernames in the Authentication
table.
Table 143: Ignore Plus (+) Character in Username Parameters
Enable The plus (+) character is ignored when attempting to match a username in the
authentication table.
Disable The plus (+) character is not ignored when attempting to match a username in the
authentication table.
3. Select whether or not the pound character (#) must be escaped in the username part of a SIP URI
in the Escape Pound (#) in SIP URI Username drop-down menu.
Table 144: Escape Pound Parameters
Enable The Pound character (#) is escaped in the username

part of a SIP URI.
Disable The Pound character (#) is not escaped in the
username part of a SIP URI.
Note that RFC 3261 specifies that the pound character
(#) needs to be escaped in the username part of a SIP
URI.
4. Select the format of the escaped characters to be used in all SIP headers in the Escape Format
drop-down menu.
Table 145: Escape Format Parameters
Lower Hexadecimal Escaped characters are displayed in a lowercase

hexadecimals format.
Upper Hexadecimal Escaped characters are displayed in a uppercase
hexadecimals format.
Additional Interop Parameters
The interop parameters allow the Aastra unit to properly work, communicate, or connect with specific IP
devices.
Call Waiting Private Number Criteria for SIP INFO

You can specify the call waiting criteria, in the form of a regular expression, that defines a private number
received in a SIP INFO.

Chapter 26 - Interop Parameters Additional Interop Parameters
 To set the Call Waiting Private Number Criteria:

1. In the sipEpMIB, set the Call Waiting Private Number Criteria in the
InteropCallWaitingSipInfoPrivateNumberCriteria variable.
sipEp.interopCallWaitingSipInfoPrivateNumberCriteria="Value"
For example, the value "(Anonymous|anonymous)" would define a calling number that is either
"Anonymous" or "anonymous" as private. The regular expression symbols to match the beginning
and end of the number are implicit and do not need to be specified. See “Regular Expressions” on
The variable is effective only if the Default Hook-Flash Processing parameter of the SIP > Misc page
is set to TransmitUsingSignalingProtocol (see “General Configuration” on page 417 for more
details).
Max-Forwards Header
Max-Forwards serves to limit the number of hops a request can make on the way to its destination. It consists
of an integer that is decremented by one at each hop. If the Max-Forwards value reaches 0 before the request
reaches its destination, it is rejected with a “483 (Too Many Hops)” error response. The Max-Forwards SIP
header is always present and the default value is 70.
Direction Attributes in a Media Stream

The Aastra unit allows you to define various direction attributes pertaining to the media stream.
When Putting a Call on Hold

Protocol (SDP)
The Aastra unit can provide the direction attribute and the meaning of the connection address “0.0.0.0” sent
in the SDP when an endpoint is put on hold.
Table 146: Direction Attributes
inactive The stream is put on hold by marking it as inactive. This is the default value. This setting
should be used for backward compatibility issues.
sendonly The stream is put on hold by marking it as sendonly. This method allows the Aastra unit to
be in conformance with RFC 3264.
 To define the direction attribute when putting a call on hold:

1. In the sipEpMIB, set the interopOnHoldSdpStreamDirection variable to the proper value.
sipEp.interopOnHoldSdpStreamDirection="Value"
Table 147: Direction Attributes Values
Value Meaning
100 inactive

Table 147: Direction Attributes Values (Continued)
Value Meaning
200 sendonly
This configuration has no effect if the interopSdpDirectionAttributeEnable variable is set to

disable (see “Direction Attribute” on page 199 for more details).
Direction Attribute
• RFC 3264: An Offer/Answer Model with Session Description
Protocol (SDP)
You can define if the SDP direction attribute is supported by the unit.
This variable applies only when the negotiated media uses an IPv4 address. The application always behaves
as if this variable is set to Enable for media using an IPv6 address.
Table 148: SDP Direction Attribute
disable No direction attribute is present in the SDP sent by the Aastra unit.
The Aastra unit ignores any direction attribute found in the SDP received from the peer.
The method to put a session on hold is in conformance with RFC 2543.
enable The Aastra unit always sends the direction attribute in the SDP of an initiated call. For
all other SDP messages sent by the unit, refer to “Enable/Disable SDP Detect Peer
Direction Attribute Support” on page 199.
If present in the SDP, the direction attribute is preferred over the connection address to
transmit session modification information.
 To define if the direction attribute is present:

1. In the sipEpMIB, set the interopSdpDirectionAttributeEnable variable to the proper value.
sipEp.interopSdpDirectionAttributeEnable="Value"
Table 149: SDP Direction Attribute
Value Meaning
0 disable
1 enable
Enable/Disable SDP Detect Peer Direction Attribute Support

You can define if the SDP direction attribute support should be autodetected in the SDP received from the
peer.
This variable is used only when the negotiated media uses an IPv4 address and when the
interopSdpDirectionAttributeEnable is enabled (see “Direction Attribute” on page 199 for more details).
The application always behaves as if this variable is set to 'Disable' for media using an IPv6 address.


Table 150: SDP Detect Peer Direction Attribute Parameters
disable The Aastra unit always sends the direction attribute in the SDP without autodetection of
peer support.
enable The initial handshake determines if the peer supports the direction attribute. The
direction attribute will be present when the peer supports it.
 To define if the SDP detect peer direction attribute is enabled or disabled:

1. In the sipEpMIB, set the interopSdpDetectPeerDirectionAttributeSupportEnable variable to
the proper value.
sipEp.interopSdpDetectPeerDirectionAttributeSupportEnable="Value"
Table 151: SDP Detect Peer Direction Attribute Values
Value Meaning
0 disable
1 enable
On Hold SDP Connection Address

You can define the value of the connection address sent in the SDP when an endpoint is on hold and no longer
listening to media packets.
This variable is used only when the negotiated media uses an IPv4 address. The application always behaves
as if this variable is set to 'MediaAddress' for media using an IPv6 address.
Table 152: On Hold SDP Connection Address Parameters
HoldAddress The connection address sent in the SDP is '0.0.0.0' if the media uses an IPv4 address.
This method is described by RFC 2543.
MediaAddress The connection address sent in the SDP is the listening address.
 To define the on hold SDP connection address:

1. In the sipEpMIB, set the interopOnHoldSdpConnectionAddress variable to the proper value.
sipEp.interopOnHoldSdpConnectionAddress="Value"
Table 153: On Hold SDP Connection Address Values
Value Meaning
100 HoldAddress
200 MediaAddress

Answering a Hold Offer with the Direction Attribute “sendonly”

Protocol (SDP)
You can define how to set the direction attribute in the SDP when answering a hold offer with the direction
attribute 'sendonly'.
inactive The stream is marked as inactive and if the stream uses an IPv4
address, the connection address is set according to the
InteropOnHoldSdpConnectionAddress variable (“On Hold SDP
Connection Address” on page 200).
revconly If the stream is currently active or receive only, it is marked as recvonly
and the connection address is set to the IP address of the unit.
If the stream is currently send only or inactive, it is marked as inactive
and the connection address is set according to the
InteropOnHoldSdpConnectionAddress variable (“On Hold SDP
Connection Address” on page 200).
 To define the behaviour with the “sendonly” direction attribute:

1. In the sipEpMIB, set the InteropOnHoldAnswerSdpStreamDirection variable to the proper value.
sipEp.interopOnHoldAnswerSdpStreamDirection="Value"
Value Meaning
100 inactive
200 Recvonly
In both cases, no direction attribute is present in the SDP if the

interopSdpDirectionAttributeEnable variable is set to disable (see “Direction Attribute” on
SDP Direction Attribute Level

Protocol (SDP)
You can define the preferred location where the stream direction attribute is set.
Table 156: SDP Direction Attribute Level
MediaOrSessionLevel If every media have the same direction, the stream direction attribute is
only present at session level.
Otherwise, the stream direction attribute is only present at media level.

Table 156: SDP Direction Attribute Level (Continued)
MediaAndSessionLevel If every media have the same direction, the stream direction attribute is
present both at session level and media level.
Otherwise, the stream direction attribute is only present at media level.
 To define the SDP direction attribute level:

1. In the sipEpMIB, set the InteropSdpDirectionAttributeLevel variable to the proper value.
sipEp.InteropSdpDirectionAttributeLevel="Value"
Table 157: SDP Direction Attribute Level
Value Meaning
100 MediaOrSessionLevel
200 MediaAndSessionLevel
Local Ring Behaviour on Provisional Response

You can set the Aastra unit so that it starts or not the local ring upon receiving a “18x Provisional” response
without SDP.
This setting does not affect the behaviour when the “18x Provisional” response contains SDP, which allows
establishing an early media session before the call is answered.
This variable does not affect the behaviour in case the '18x Provisional' response contains SDP, in which case
the media stream, if present, is played.
Figure 82: Local Ring Behaviour
Disable The local ring is not started on a '18x Provisional' response without SDP, with one
exception: the '180 Ringing' without SDP will start the local ring if the media stream is
not already established.
LocalRingWhenNo : The local ring is started on any '18x Provisional' response without SDP if the media
EstablishedMediaS stream is not already established.
tream
LocalRingAlways The local ring is always started on any '18x Provisional' response without SDP.
 To define the local ring behaviour on provisional response:

1. In the sipEpMIB, set the interopLocalRingOnProvisionalResponse variable to the proper value.
sipEp.interopLocalRingOnProvisionalResponse="Value"
Figure 83: Local Ring Values
Value Meaning
0 disable
1 LocalRingWhenNoEstablishedMediaStream

Figure 83: Local Ring Values (Continued)
Value Meaning
2 LocalRingAlways
Session ID and Session Version Number in the Origin Field of the SDP
You can define the maximum length of the session ID and the session version number in the origin line (o=)
of the SDP. This allows the Aastra unit to be compatible with 3rd party vendor equipment.
Table 158: Maximum Length Parameters
Length Description
max-32bits The session ID and the session version number are represented with a 32 bit integer.
They have a maximum length of 10 digits.
max-64bits The session ID and the session version number are represented with a 64 bit integer.
They have a maximum length of 20 digits. This is the default value.
 To set the maximum length of the session ID and the session version number:
1. In the sipEpMIB, set the interopSdpOriginLineSessionIDAndVersionMaxLength variable with
the proper length.
sipEp.interopSdpOriginLineSessionIdAndVersionMaxLength="Value"
Table 159: Maximum Length Values
Value Meaning
100 max-32bits
200 max-64bits
Register Home Domain Override

By default, the address-of-record in the “To” header uses the value set in the Proxy Host field of the SIP/
Configuration page for the host/port part. See “SIP Servers Configuration” on page 282 for more details. You
can override this value if required.
 To override the register home domain value:

1. In the sipEpMIB, set the interopRegisterHomeDomainOverride variable with the override home
domain value.
sipEp.interopRegisterHomeDomainOverride="IP_Address"
The address of record in the register will use this string instead of the home domain proxy. If the
variable is empty, the value of the Proxy Host field is used.
The host is also overridden in the From and Call-Id headers since they match the To header.

DNS SRV Record Lock

Standards Supported • RFC 3263 - Session Initiation Protocol (SIP): Locating SIP
Servers
You can configure the Aastra unit to always use the same DNS SRV record for a SIP call ID. As a result, a call
or registration always uses the same destination until the destination is unreachable or the unit receives a
different DNS SRV result.
Table 160: DNS SRV Record Lock Parameters
Length Description
disable The behaviour follows RFC 3263.

enable All messages during a call or registration use the same SRV record.
 To enable the DNS SRV record lock feature:

1. In the sipEpMIB, set the interopLockDnsSrvRecordPerCallEnable variable to enable.
sipEp.interopLockDnsSrvRecordPerCallEnable="Value"
Figure 84: DNS SRV Record Lock Values
Value Meaning
0 disable
1 enable
Listening for Early RTP

You can set the Aastra unit so that it listens for RTP before the reception of a response with SDP. This feature
only applies to calls initiated from analog endpoints (FXS/FXO) with non-secure RTP.
Table 161: Early RTP Parameters
Length Description
enable The RTP port is opened after the initial INVITE has been sent, without waiting for a provisional or
final response with SDP to be received. No local ring is generated. This conforms to section 5.1 of
RFC 3264.
disable The RTP port is opened only after a response with SDP is received.
Warning: Do not enable this feature unless the server supports early RTP (or early media). Failing so
prevents any ringing to be heard for outgoing calls.
 To enable the Early RTP feature:

1. In the sipEpMIB, set the InteropListenForEarlyRtpEnable variable to enable.
sipEp.InteropListenForEarlyRtpEnable="Value"


Figure 85: Early RTP Values
Value Meaning
0 disable
1 enable
Resolve Route Header

The Aastra unit has a parameter that allows you to resolve the FQDN in the top-most route header of outgoing
packets.
Table 162: Resolve Route Header Parameters
Length Description
enable The FQDN in the top-most route header is replaced by the IP address of the packet's destination
if the FQDN matches the gateway's configured outbound proxy.
disable The route header is not modified.
 To resolve the route header:

1. In the sipEpMIB, set the InteropResolveRouteHeaderEnable variable with the proper value.
sipEp.interopResolveRouteHeaderEnable="Value"
Figure 86: Resolve Route Header Values
Value Meaning
0 disable
1 enable
ACK Branch Matching

You can configure the method used to match incoming ACK SIP packets.
Table 163: ACK Branch Matching Parameters
Rfc3261 Follows the method described in RFC 3261 (section 8.1.1.7). The branch value in the topmost
via of the ACK request to a 2XX response MUST be different than the one of the INVITE.
Rfc3261Wi Follows the method described in RFC 3261 (section 8.1.1.7) but enables the handling of ACK
thoutAck requests (for 2XX responses) that have the same branch value in the topmost via as the
INVITE.
 To set ACK branch matching:

1. In the sipEpMIB, set the interopAckBranchMatching variable with the proper value.
sipEp.interopAckBranchMatching="Value"


Figure 87: ACK Branch Matching Values
Value Meaning
100 Rfc3261
200 Rfc3261WithoutAck
Ignore Require Header

You can define whether or not the Require Header must be ignored when processing the incoming SIP Client
requests (INVITE, re-INVITE, Bye, etc.).
Table 164: Ignore Require Header Parameters
Enable The Require Header is ignored and no validation about these options-tags is performed.
Disable The Require Header options-tags are validated and, when an option-tag is not supported, a
420 (Bad Extension) response is sent.
The supported options-tags are:
• * 100rel
• * replaces
• * timer
 To set whether or not to ignore the Require header:

1. In the sipEpMIB, set the interopIgnoreRequireHeaderEnable variable with the proper value.
sipEp.interopIgnoreRequireHeaderEnable="Value"
Figure 88: Ignore Require Header Values
Value Meaning
0 disable
1 enable
Ringing Response Code

You can define the response code sent back when the endpoint starts ringing.
Table 165: Ringing Response Code Parameters
Send180Ringing Sends out a '180 Ringing' response without a body.

Send183WithSdp Returns a '183 Session Progress' packet with SDP (needed if the endpoint is
required to generate ringback on connection).

 To set the ringing response code:

1. In the sipEpMIB, set the InteropRingingResponseCode variable with the proper value.
sipEp.InteropRingingResponseCode="Value"
Figure 89: Ringing Response Code Values
Value Meaning
100 Send180Ringing
200 Send183WithSdp
Reject Code for No Resource

You can define the rejection code used when all lines of the group are unavailable.
Table 166: Reject Code for No Resource Parameters
TemporarilyUnavailable The '480 Temporarily Unavailable' rejection code is used.

BusyHere The '486 Busy Here' rejection code is used.
 To set the reject code:

1. In the sipEpMIB, set the InteropRejectCodeForNoRessource variable with the proper value.
sipEp.InteropRejectCodeForNoRessource="Value"
Figure 90: Reject Code Values
Value Meaning
100 TemporarilyUnavailable
200 BusyHere
Reject Code for Unsupported SDP Offer

You can define the rejection code used when an offer is received with invalid or unsupported SDP Offer. RFC
3261 recommends using the error code 488 'Not Acceptable Here'.
Table 167: Reject Code for Unsupported SDP Offer Parameters
UnsupportedMediaType The SIP error code 415 'Unsupported Media Type' is returned if the Content-
Type is invalid; the payload is missing or the SDP content is invalid.
NotAcceptableHere The SIP error code 488 'Not Acceptable Here' is returned if the SDP content is
invalid.

 To set the reject code:

1. In the sipEpMIB, set the InteropRejectCodeForUnsupportedSdpOffer variable with the proper
value.
sipEp.InteropRejectCodeForUnsupportedSdpOffer="Value"
Figure 91: Reject Code Values
Value Meaning
415 UnsupportedMediaType
488 NotAcceptableHere
SIP User-Agent Header Format

You can define the text to display in the SIP User-Agent header. You can use macros to include information
specific to the unit.
You can also define the same information in the HTTP User-Agent header. See “HTTP User-Agent Header
Format” on page 42 for more details.
 To set the SIP User-Agent header format:

1. In the sipEpMIB, set the User-Agent header format in the interopUaHeaderFormat variable.
sipEp.interopUaHeaderFormat="Value"
where Value may contain any text, as well as one or more of the following macros:
Table 168: Macros Supported
Macro Description
%version% Application version.

%mac% MAC address.
%product% Product name.
%profile% Profile.
%% Insert the % character.
For instance, the default value is:

%product%/v%version% %profile%
SIP INFO Without Content Answer

You can define the response of the Aastra unit to a received SIP INFO with no message body for an existing
call.
RFC 2976 recommends that a 200 OK response MUST be sent for an INFO request with no message body if
the INFO request was successfully received for an existing call.
Table 169: Reject Code for Unsupported SDP Offer Parameters
UnsupportedMediaType The unit responds with the SIP error code 415 'Unsupported Media Type'.
Ok The unit responds with a 200 OK.

 To define the SIP INFO Without Content Answer behaviour:

1. In the sipEpMIB, set the interopSipInfoWithoutContentAnswer variable with the proper
behaviour.
sipEp.interopSipInfoWithoutContentAnswer="Value"
Table 170: SIP INFO Values
Value Meaning
200 Ok
415 UnsupportedMediaType
Unsupported Content-Type
You can define the behaviour of the Aastra unit upon reception of a SIP packet containing multiple
unsupported Content-Type in the payload.
Table 171: Unsupported Content-Type Parameters
Reject Unsupported Content-Type are rejected.

Allow Unsupported Content-Type are allowed and ignored if at least one Content-Type
is supported.
Ignore Unsupported Content-Type are ignored.
Note: When ignored, unsupported Content-Type are treated as if they were not present in the packet.
 To define the unsupported Content-Type behaviour:

1. In the sipEpMIB, set the interopUnsupportedContentType variable with the proper behaviour.
sipEp.interopUnsupportedContentType="Value"
Table 172: Unsupported Content-Type Values
Value Meaning
100 Reject
200 Allow
300 Ignore


C H A P T E R
27 Miscellaneous SIP Parameters
This chapter describes miscellaneous SIP parameters you can set:

 SIP penalty box parameters
 How to override the default mapping of error causes defined in RFC 3398.
 Additional Headers
 PRACK
 Session Refresh
 SIP Gateway Configuration
 SIAastraP Blind Transfer Method
 Diversion Configuration
 DNS Configuration
 Event Handling Configuration
 Messaging Subscription
SIP Penalty Box
The penalty box feature is used when a given host FQDN resolves to a non-responding address. When the
address times out, it is put into the penalty box for a given amount of time. During that time, the address in
question is considered as “non-responding” for all requests.
This feature is most useful when using DNS requests returning multiple or varying server addresses. It makes
sure that, when a host is down, users wait a minimal amount of time before trying a secondary host.
When enabled, this feature takes effect immediately on the next call attempt.
The penalty box feature is applied only when using UDP or TCP connections established with a FQDN. A
similar penalty box feature for the TLS persistent connections is available via the TLS Persistent Retry Interval
parameter. See “SIP Transport Type” on page 305 for more details.
Penalty Box vs Transport Types

Aastra recommends to use this feature with care when supporting multiple transports (see “Chapter 28 - SIP
Transport Parameters” on page 305 for more details) or you may experience unwanted behaviours.
When the Aastra unit must send a packet, it retrieves the destination from the packet. If the destination address
does not specify a transport to use and does not have a DNS SRV entry that configures which transport to
use, then the Aastra unit tries all transports it supports, starting with UDP. If this fails, it tries with TCP. The
unit begins with UDP because all SIP implementations must support this transport, while the mandatory
support of TCP was only introduced in RFC 3261.
Note: It is not the destination itself that is placed in the penalty box, but the combination of address, port
and transport. When a host is in the penalty box, it is never used to try to connect to a remote host unless it
is the last choice for the Aastra unit and there are no more options to try after this host.
Let’s say for instance that the Aastra unit supports both the UDP and TCP transports. It tries to reach endpoint
“B” for which the destination address does not specify a transport and there is no DNS SRV entry to specify
which transports to use in which order. It turns out that this endpoint “B” is also down. In this case, the Aastra
unit first tries to contact endpoint “B” via UDP. After a timeout period, UDP is placed in the penalty box and the
unit then tries to contact endpoint “B” via TCP. This fails as well and TCP is also placed in the penalty box.

Chapter 27 - Miscellaneous SIP Parameters Error Mapping
Now, let’s assume endpoint “B” comes back to life and the Aastra unit tries again to contact it before UDP and
TCP are released from the penalty box. First, the unit tries UDP, but it is currently in the penalty box and there
is another transport left to try. The Aastra unit skips over UDP and tries the next target, which is TCP. Again,
TCP is still in the penalty box, but this time, it is the last target the Aastra unit can try, so penalty box or not,
TCP is used all the same to try to contact endpoint “B”.
There is a problem if endpoint “B” only supports UDP (RFC 2543-based implementation). Endpoint “B” is up,
but the Aastra unit still cannot contact it: with UDP and TCP in the penalty box, the unit only tries to contact
endpoint “B” via its last choice, which is TCP.
The same scenario would not have any problem if the penalty box feature was disabled. Another option is to
disable TCP in the Aastra unit, which makes UDP the only possible choice for the unit and forces to use UDP
even if it is in the penalty box.
You must fully understand the above problem before configuring this feature. Mixing endpoints that do not
support the same set of transports with this feature enabled can lead to the above problems, so it is suggested
to either properly configure SRV records for the hosts that can be reached or be sure that all hosts on the
network support the same transport set before enabling this feature.
Penalty Box Configuration

The following steps describe how to configure the penalty box feature.
 To set the SIP penalty box parameters:

1. In the web interface, click the SIP link, then the Misc sub-link.
Figure 92: SIP Configuration – Misc Web Page
2
3
2. In the Penalty Box section, enable the SIP penalty box feature by selecting Enable in the Penalty
Box Activation drop-down menu.
The penalty box is always “active”. This means that even if the feature is disabled, IP addresses are
marked as invalid, but they are still tried. This has the advantage that when the feature is enabled,
IP addresses that were already marked as invalid are instantly put into the penalty box.
3. Set the amount of time, in seconds, that a host spends in the penalty box in the Penalty Box Time
field.
Changing the value does not affect IP addresses that are already in the penalty box. It only affects
new entries in the penalty box.
Error Mapping
Standards Supported • RFC 3398: Integrated Services Digital Network (ISDN) User
Part (ISUP) to Session Initiation Protocol (SIP) Mappinga
a. Only the ISDN to SIP error mapping is supported.
You can override the default mapping of error causes defined in RFC 3398.The web interface offers two
sections:
 The SIP To Cause Error Mapping section allows you to override the default mapping for SIP

Error Mapping Software Configuration Guide
code to ISDN cause.

 The Cause To SIP Error Mapping section allows you to override the default mapping for ISDN
cause to SIP code.
The following standard SIP codes are available:
400: Bad Request 414: Request-URI too long 485: Ambiguous

401: Unauthorized 415: Unsupported media type 486: Busy here
402: Payment required 416: Unsupported URI Scheme 500: Server internal error
403: Forbidden 420: Bad extension 501: Not implemented
404: Not found 421: Extension Required 502: Bad gateway
405: Method not allowed 423: Interval Too Brief 503: Service unavailable
406: Not acceptable 480: Temporarily unavailable 504: Server time-out
407: Proxy authentication 481: Call/Transaction Does not 504: Version Not Supported
required Exist 513: Message Too Large
408: Request timeout 482: Loop Detected 600: Busy everywhere
410: Gone 483: Too many hops 603: Decline
413: Request Entity too long 484: Address incomplete 604: Does not exist anywhere
You can also map any other custom code between 400 and 699.

The following standard ISDN cause numbers specified in Q.931 are available:
Normal event: Service or option not implemented:

1: Unassigned (unallocated) number. 65: Bearer capability not implemented.
2: No route to specified transit network. 66: Channel type not implemented.
3: No route to destination. 69: Requested facility not implemented.
6: Channel unacceptable. 70: Only restricted digital information bearer.
7: Call awarded and being delivered in an 79: Service or option not implemented,
established channel. unspecified.
17: User busy.
18: No user responding. Invalid Message
19: User alerting, no answer. 81: Invalid call reference value.
20: Subscriber absent. 82: Identified channel does not exist.
21: Call rejected. 83: A suspended call exists, but this call identity
22: Number changed. does not.
23: Redirection to new destination. 84: Call identity in use.
26: Non-selected user clearing. 85: No call suspended.
27: Destination out of order. 86: Call having the requested call identity has been
cleared.
28: Invalid number format (incomplete number).
87: user not member of CUG.
29: Facility rejected.
88: Incompatible destination.
30: Response to STATUS ENQUIRY.
91: Invalid transit network selection.
31: Normal, unspecified.
95: Invalid message, unspecified.
Resource unavailable:
Protocol error
34: No circuit/channel available.
96: Mandatory information element is missing.
38: Network out of order.
97: Message type non-existent or not
41: Temporary failure.
implemented.
42: Switching equipment congestion.
98: Message not compatible with call state or
43: Access information discarded. message type non-existent or not implemented.
44: Requested circuit/channel not available. 99: Information element non-existent or not
47: Resource unavailable, unspecified. implemented.
100: Invalid information element contents.
Service or option not available: 101: Message not compatible with call state.
55: Incoming calls barred within CUG. 102: Recovery on time expiry.
57: Bearer capability not authorized. 111: Protocol error, unspecified.
58: Bearer capability not presently available. Interworking
63: Service or option not available, unspecified. 127: Interworking, unspecified
You can also map any other custom code between 1 and 127.
SIP to Cause Error Mapping

This section describes how to override the default mapping of ISDN error causes.

 To override the default mapping of ISDN error causes:

1. In the SIP To Cause Error Mapping section of the Misc page, click the button to add a new row.
Figure 93: SIP To Cause Error Mapping Section
This brings you to the Configure New SIP To Cause Error Mapping panel.
2. Enter the SIP code in the SIP Code field, then the corresponding ISDN cause number in the Cause
column.
You can use the Suggestion column’s drop-down menu to select between available code values.
Figure 94: Configure New SIP To Cause Error Mapping Panel
3. Click Submit.
This brings you back to the main Misc web page.
You can delete an existing row by clicking the button.
You can modify the Cause value by typing a new code in the field. See “SIP To Cause Default Error
Mapping” on page 215 for the default mappings as per RFC 3398.
SIP To Cause Default Error Mapping

Table 173 lists the default mappings as per RFC 3398.
Table 173: SIP To Cause Default Error Mapping
SIP Response Received Cause Value
400 Bad Request 41 Temporary Failure

401 Unauthorized 21 Call rejected
402 Payment required 21 Call rejected
403 Forbidden 21 Call rejected
404 Not found 1 Unallocated number
405 Method not allowed 63 Service or option unavailable
406 Not acceptable 79 Service/option not implemented
407 Proxy authentication required 21 Call rejected
408 Request timeout 102 Recovery on timer expiry
410 Gone 22 Number changed (w/o diagnostic)
413 Request Entity too long 127 Interworking
414 Request-URI too long 127 Interworking

Table 173: SIP To Cause Default Error Mapping (Continued)
SIP Response Received Cause Value
415 Unsupported media type 79 Service/option not implemented

416 Unsupported URI Scheme 127 Interworking
420 Bad extension 127 Interworking
421 Extension Required 127 Interworking
423 Interval Too Brief 127 Interworking
480 Temporarily unavailable 18 No user responding
481 Call/Transaction Does not Exist 41 Temporary Failure
482 Loop Detected 25 Exchange - routing error
483 Too many hops 25 Exchange - routing error
484 Address incomplete 28 Invalid Number Format
485 Ambiguous 1 Unallocated number
486 Busy here 17 User busy
500 Server internal error 41 Temporary failure
501 Not implemented 79 Not implemented, unspecified
502 Bad gateway 38 Network out of order
503 Service unavailable 41 Temporary failure
504 Server time-out 102 Recovery on timer expiry
504 Version Not Supported 127 Interworking
513 Message Too Large 127 Interworking
600 Busy everywhere 17 User busy
603 Decline 21 Call rejected
604 Does not exist anywhere 1 Unallocated number
Cause to SIP Error Mapping

This section describes how to override the default mapping of SIP codes.
 To override the default mapping of SIP codes:

1. In the Cause To SIP Error Mapping section of the Misc page, click the button to add a new row.
Figure 95: Cause To SIP Error Mapping Section
This brings you to the Configure New Cause To SIP Error Mapping panel.
2. Enter the ISDN cause number in the Cause column, then the corresponding SIP code in the SIP
Code field.
You can use the Suggestion column’s drop-down menu to select between available code values.

Figure 96: Configure New Cause To SIP Error Mapping Panel
3. Click Submit.
This brings you back to the main Misc web page.
You can delete an existing row by clicking the button.
You can modify the SIP Code value by typing a new code in the field. See “Cause To SIP Default
Error Mapping” on page 217 for the default mappings as per RFC 3398.
Cause To SIP Default Error Mapping

Table 174 lists the default mappings as per RFC 3398.
Table 174: Cause To SIP Default Error Mapping
ISUP Cause Value SIP Response
Normal Event
1 unallocated number 404 Not Found
2 no route to network 404 Not Found
3 no route to destination 404 Not Found
16 normal call clearing --- BYE or CANCEL
17 user busy 486 Busy Here
18 no user responding 408 Request Timeout
19 no answer from the user 480 Temporarily unavailable
20 subscriber absent 480 Temporarily unavailable
21 call rejected 403 Forbidden
22 number changed (w/o diagnostic) 410 Gone
22 number changed (w/ diagnostic) 301 Moved Permanently
23 redirection to new destination 410 Gone
26 non-selected user clearing 404 Not Found
27 destination out of order 502 Bad Gateway
28 address incomplete 484 Address incomplete
29 facility rejected 501 Not implemented
31 normal unspecified 480 Temporarily unavailable
Resource Unavailable
34 no circuit available 503 Service unavailable
38 network out of order 503 Service unavailable

Table 174: Cause To SIP Default Error Mapping (Continued)
ISUP Cause Value SIP Response
41 temporary failure 503 Service unavailable

42 switching equipment congestion 503 Service unavailable
47 resource unavailable 503 Service unavailable
Service or Option not Available
55 incoming calls barred within CUG 403 Forbidden
57 bearer capability not authorized 403 Forbidden
58 bearer capability not presently available 503 Service unavailable
Service or Option not Implemented
65 bearer capability not implemented 488 Not Acceptable Here
70 only restricted digital available 488 Not Acceptable Here
79 service or option not implemented 501 Not implemented
Invalid message
87 user not member of CUG 403 Forbidden
88 incompatible destination 503 Service unavailable
Protocol error
102 recovery of timer expiry 504 Gateway timeout
111 protocol error 500 Server internal error
Interworking
127 interworking unspecified 500 Server internal error

Additional Headers Software Configuration Guide
Additional Headers
You can define whether or not the Aastra unit uses additional SIP headers.
 To use additional SIP headers:

1. In the Additional Headers section of the Misc page, select the method to use in the Reason Header
Support drop-down menu.
Figure 97: Reason Header Section
1
2
Table 175: Reason Header Support Parameters
None Silently ignores any incoming reason headers and does not
send the reason header.
SendQ850 Silently ignores incoming reason codes and sends the SIP
reason code when the original Q.850 code is available. The
reason code sent is not affected by the entries in the Error
Mapping SIP To Cause table.
ReceiveQ850 Uses the incoming Q.850 reason cause header. When
received, the reason code supersedes any entrie s in the
Error Mapping SIP To Cause table.
SendReceiveQ850 Uses the incoming Q.850 reason cause header and sends
the SIP reason code when the original Q.850 code is
available. When received, the reason code supersedes any
entries in the Error Mapping SIP To Cause table. The
reason code sent is not affected by the entries in the Error
Mapping SIP To Cause table.
2. Select how the Referred-By header is used when participating in a transfer in the Referred-By
Table 176: Referred-By Support Parameters
None When acting as the transferor (sending the REFER), the

REFER does not contain a Referred-By header.
When acting as the transferee (receiving the REFER and
sending the INVITE to the target), the Referred-By header is
not copied from the REFER to the INVITE.
HeaderOnly When acting as the transferor (sending the REFER), the
Referred-By header contains the SIP URI of the transferor.
When acting as the transferee (receiving the REFER and
sending the INVITE to the target), the Referred-By header is
copied from the REFER to the INVITE.

4. Set the interval, in seconds, at which SIP Keep Alive requests using SIP OPTIONS or Ping are sent
to verify the server status in the Keep Alive Interval field.

Chapter 27 - Miscellaneous SIP Parameters PRACK
PRACK
Standards Supported • RFC 3262: Reliability of Provisional Responses in the

Session Initiation Protocol (SIP)
• RFC 3311: The Session Initiation Protocol (SIP) UPDATE
Methoda
a. Only support receiving UPDATE. Sending an UPDATE is not supported.
The Aastra unit supports reliable provisional responses (PRACK) as per RFC 3262. You can define this
support when acting as a user agent client and when acting as a user agent server.
The Aastra unit supports the UPDATE as per RFC 3311; however, its support is limited to reception.
 To define the PRACK support:

1. In the PRACK section of the Misc page, define the support of RFC 3262 (PRACK) when acting as
a user agent server in the UAS PRACK Support drop-down menu.
Figure 98: PRACK Section
1
2
Table 177: PRACK User Agent Server Parameters
Unsupported The option tag “100rel” is ignored if present in the Supported

or Required header of received initial INVITEs and
provisional responses are not sent reliably as per RFC 3261.
Supported If the option tag “100rel” is present in the Supported or
Required header of initial received INVITEs, provisional
responses are sent reliably as per RFC 3262 by adding the
option tag “100rel” to the Require header.
Receiving an UPDATE request to negotiate “early media” is supported only if you have selected
Supported.
2. Define the support of RFC 3262 (PRACK) when acting as user agent client in the UAC PRACK
Table 178: PRACK User Agent Client Parameters
Unsupported The option tag “100rel” is not added in the Supported or

Required header of sent INVITEs as per RFC 3261. If the
provisional response contains a Require header field with
the option tag “100rel”, the indication is ignored and no
PRACK are sent.
Supported The option tag “100rel” is added to the Supported header of
sent initial INVITEs as per RFC 3262. If the received
the option tag “100rel”, the response is sent reliably using
the PRACK method.

Session Refresh Software Configuration Guide
Table 178: PRACK User Agent Client Parameters (Continued)
Required The option tag “100rel” is added to the Require header of

sent initial INVITEs as per RFC 3262. If the received
the option tag “100rel”, the response is sent reliably using
the PRACK method.
Forked Provisional Responses Behaviour

You can configure the unit's behaviour when receiving forked provisional answers. This configuration has no
effect if the UAC PRACK Support drop-down menu is set to a value other than Unsupported.
Table 179: Forked Provisional Responses Behaviour Parameters
Value Description
InterpretFirst Only the first provisional answer is interpreted. Following responses do not change the
state of the call and the SDP is ignored if present.
InterpretAll Each forked provisional response received by the unit is interpreted replacing the previous
one. If the response contains SDP, it replaces previous answers if any.
 To set the forked provisional responses behaviour:

1. In the sipEpMIB, define the behaviour in the interopForkedProvisionalResponsesBehavior
variable.
sipEp.interopForkedProvisionalResponsesBehavior=[value]
Table 180: Forked Provisional Responses Behaviour Values
Value Meaning
100 InterpretFirst
200 InterpretAll
Session Refresh
This section allows you to define session refresh and session timers parameters. Session timers apply to the
whole unit.

Chapter 27 - Miscellaneous SIP Parameters Session Refresh
 To set Session Refresh information:

1. In the Session Refresh section of the Misc page, define whether to enable or disable the session
expiration services in the Session Refresh Timer Enable drop-down menu.
Figure 99: Session Refresh Section
1
2
1
2
Disabling this service is not recommended since it will make 'dead' calls impossible to detect.
See “Background Information” on page 222 for more details.
2. Set the session timer minimum expiration delay, in seconds, in the Minimum Expiration Delay (s)
field.
This is the minimum value, in seconds, for the periodical session refreshes. It must be equal to or
smaller than the maximum value. This value is reflected in the Min-SE header.
The Min-SE value is a threshold under which proxies and user agents on the signalling path are not
allowed to go. Increasing the minimum helps to reduce network traffic, but also makes “dead” calls
longer to detect.
3. Set the session timer maximum expiration delay, in seconds, in the Maximum Expiration Delay (s)
field.
This is the suggested maximum time, in seconds, for the periodical session refreshes. It must be
equal to or greater than the minimum value. This value is reflected in the Session-Expires header.
Increasing the maximum helps to reduce network traffic, but also makes “dead” calls longer to
detect.
Note: When the Maximum Expiration Delay value is lower than the Minimum Expiration Delay value, the
minimum and maximum expiration delay values in INVITE packets are the same as the value set in the
Minimum Expiration Delay field.
4. Select the method used for sending Session Refresh Requests in the Use UPDATE for Session
Refresh parameter.
Table 181: UPDATE for Session Refresh Parameters
ReInvite Session Refresh Requests are sent with the INVITE method.
Update Session Refresh Requests are sent with the UPDATE
method.
Session Refresh Requests can be received via both methods, regardless of how this parameter is
configured.
Background Information
The following explains how the session timers are used.
What is the session timer extension?

The session timer extension allows detecting the premature end of a call caused by a network problem or a
peer’s failure by resending a refresh request at every n seconds. This refresh request is either an reINVITE or
an UPDATE, according to the configuration of the Session Refresh Request Method parameter (see “PRACK”
on page 220).

SIP Gateway Configuration Software Configuration Guide
A successful response (200 OK) to this refresh request indicates that the peer is still alive and reachable. A
timeout to this refresh request may mean that there are problems in the signalling path or that the peer is no
longer available. In that case, the call is shut down by using normal SIP means.
SDP in Session Timer reINVITEs or UPDATEs

The reINVITE is sent with the last SDP that was negotiated. Receiving a session timer reINVITE should not
modify the connection characteristics.
If the reINVITE method is used, it is sent with the last SDP that was negotiated. Reception of a session timer
reINVITE should not modify the connection characteristics. If the UPDATE method is used, it is sent without
any SDP offer. REMPLACER
Relation Between Minimum and Maximum Values

A user agent that receives a Session-Expires header whose value is smaller than the minimum it is willing to
accept replies a “422 Timer too low” to the INVITE and terminates the call. The phone does not ring.
It is up to the caller to decide what to do when it receives a 422 to its INVITE. The Aastra unit will automatically
retry the INVITE, with a Session-Expires value equal to the minimum value that the user agent server was
ready to accept (located in the Min-SE header). This means that the maximum value as set in the Aastra unit
might not be followed. This has the advantageous effect of establishing the call even if the two endpoints have
conflicting values. The Aastra unit will also keep retrying as long as it gets 422 answers with different Min-SE
values.
Who Refreshes the Session?

Sending a session timer reINVITE or UPDATE is referred to as refreshing the session. Normally, the user
agent server that receives the INVITE has the last word on who refreshes. The Aastra unit always lets the user
agent client (caller) perform the refreshes if the caller supports session timers. In the case where the caller
does not support session timers, the Aastra unit assumes the role of the refresher.
SIP Gateway Configuration
You can define whether or not to override the SIP domain used.
 To set the SIP domain override:

1. In the SIP Gateway Configuration section of the Misc page, define whether or not to override the
SIP domain used in the SIP Domain field.
If not empty, the address of record uses this string instead of the home domain proxy (Proxy Host
field of the Servers sub-page – SIP Default Servers section (“SIP Servers Configuration” on
page 282).
Figure 100: SIP Gateway Configuration Section
1
2
SIP Blind Transfer Method
You can set the SIP transfer method when an endpoint is acting as the transferor in a blind transfer scenario.

Chapter 27 - Miscellaneous SIP Parameters Diversion Configuration
 To set the SIP blind transfer method:

1. In the SIP Transfer section of the Misc page, set the Blind Transfer Method.
Figure 101: SIP Transfer Section
Table 182: SIP Blind Transfer Method Parameters
Semi Attended When blind transfer is invoked by the transferor, the device
sends immediately a REFER (it does not wait for the
reception of the 200OK response). This allows the call
transfer to be executed before the transfer-target answers.
The transferee and the target are then connected together
early and the transferee can hear the ringback from the
target until the target answers.
Semi Attended Confirmed When blind transfer is invoked by the transferor, the device
waits for reception of the 200 OK from the transfer-target
before sending a REFER to the transferee.
Semi Attended Cancelled This method is similar to the Semi Attended Transfer
method except that the INVITE sent to the transfer-target is
cancelled when the blind transfer is invoked before
receiving a 200OK (INVITE). In case where the transferor
receives a 200OK (INVITE) from the transfer-target before
receiving of a 487 Request Terminated, the transfer stays
ongoing and it behaves as a Semi Attended Confirmed
Transfer.
Diversion Configuration
You can define call diversion parameters.
Note: The Diversion feature is not available in the NI2 and QSIG signalling protocols. See “PRI
Configuration” on page 184 for more details on how to configure the signalling protocol.
 To set the call diversion parameters:

1. In the Diversion section of the Misc page, set the Methcd drop-down menu with the SIP method
used to receive/send call diversion information in an INVITE.
The gateways available are those defined in “SIP Gateways Configuration” on page 277.
Figure 102: Diversion Configuration Section
Table 183: Diversion Parameters
None No diversion information is sent in SIP messages.

Diversion Configuration Software Configuration Guide
Table 183: Diversion Parameters (Continued)
Diversion Header The SIP gateway supports the SIP header 'Diversion' (RFC
5806) in received and sent INVITEs, as well as in 302
messages.

Chapter 27 - Miscellaneous SIP Parameters DNS Configuration
DNS Configuration
You can define DNS-related parameters.
 To set the DNS-related parameters:

1. In the DNS section of the Misc page, set the Supported DNS Queries drop-down menu with the type
of DNS queries that the SipEp service supports and uses.
Figure 103: DNS Configuration Section
Table 184: DNS Parameters
Address Sends only Address requests (type A).

SRV Sends a Service request (type SRV) first and then Address
requests (type A) if needed.
NAPTR Sends a Naming Authority Pointer request (type NAPTR)
first and then Service requests (type SRV) or Address
requests (type A) as needed.
Event Handling Configuration
The Aastra unit supports receiving event handling Notifications to start a remote reboot or a sync of
configuration for specific endpoint(s). The event handling Notifications "reboot" or "check-sync" is not
specified in an Allow-Events header. The Aastra unit supports the Notify without subscription.
It is recommended to use these event handling notifications only when the SIP transport is secure (TLS) or
when the firewall filters the requests sent to the unit.
 To set the event handling parameters:

1. In the Event Handling section of the Misc page, set the Reboot column of each available gateway
to define whether or not the SIP gateway can start a remote reboot via a SIP NOTIFY Event.
This specifies whether a remote reboot via a SIP NOTIFY message event is supported or not for a
specific SIP gateway.
Figure 104: Event Handling Parameters

1 2
Table 185: Reboot Event Handling Parameters
Rejected The "reboot" notification is rejected on reception.

Messaging Subscription Software Configuration Guide
Table 185: Reboot Event Handling Parameters (Continued)
Restart When receiving a "reboot" notification, a restart of the unit is

done.
2. Set the CheckSync column of each available gateway to define whether or not the SIP gateway can
transfer and run a configuration file via a SIP NOTIFY Event.
This specifies whether a transfer script via a SIP NOTIFY message event is supported or not for a
Table 186: CheckSync Event Handling Parameters
Rejected The "check-sync" notification is rejected on reception.

TransferScript When receiving a "check-sync" notification, the
Conf.ConfiguredScriptsTransferAndRun command is
executed.
Messaging Subscription
The Aastra unit allows you to add the username in the Request-URI of SUBSCRIBEs it sends.
 To set the messaging subscription:

1. In the Messaging Subscription section of the Misc page, set the Username in Request-URI drop-
down menu, set whether or not the unit adds the username in the request URI of MWI SUBSCRIBE
requests.
Figure 105: Messaging Subscription Parameters
Table 187: Messaging Subscription Parameters
Enable The unit adds the username in the Request-URI of sent MWI
SUBSCRIBE requests.
Disable No username in Request-URI of MWI SUBSCRIBE
requests sent by the unit.

Chapter 27 - Miscellaneous SIP Parameters Messaging Subscription

AastraMedia Parameters
C H A P T E R
28 Voice & Fax Codecs

Configuration
This chapter describes the voice and fax codec configuration parameters.
 Codec descriptions.
 How to enable and disable the codecs.
 How to set the individual codecs’ parameters.
Standards Supported • RFC 3550: RTP: A Transport Protocol for Real-Time

Applications
• RFC 3551: RTP Profile for Audio and Video Conferences with
Minimal Control
Codec Descriptions
The Aastra unit supports several voice and fax codecs. It also supports unicast applications, but not multicast
ones. All voice transport is done over UDP.
All the endpoints of the Aastra unit can simultaneously use the same codec (for instance, G.711 PCMA), or a
mix of any of the supported codecs. Set and enable these codecs for each endpoint.
Table 188: Codecs Comparison
Compression Voice Quality
G.711 None Excellent

G.723.1a Highest Good
G.726 Medium Fair
G.729a/ab High Fair/Good
a. This codec is not available on the Aastra Series models.
G.711 A-Law and µ-Law

Standards Supported • ITU-T Recommendation G.711
The audio data is encoded as 8 bits per sample, after logarithmic scaling.
Table 189: G.711 Features
Feature Description
Packetization time Range of 10 ms to 30 ms with increments of 10 ms. See “G.711 Codec

Parameters” on page 237 for more details.
For the reception, the range is extended from 10 ms to 100 ms with
increments of 1 ms only if the stream is not encrypted (SRTP).
Voice Activity Detection (VAD) Two levels of detection are available: transparent or conservative. See
“Generic Voice Activity Detection (VAD)” on page 237 for more details.

Chapter 28 - Voice & Fax Codecs Configuration Codec Descriptions
Table 189: G.711 Features (Continued)
Feature Description
Comfort noise Uses custom comfort noise as defined in RFC 3389.

Available for voice Yes
Available for fax Yes
Available for modem Yes
G.723.1
Standards Supported • ITU-T Recommendation G.723.1a
a. This codec is not available on the Aastra Series models.
Dual-rate speech coder for multimedia communications transmitting at 5.3 kbit/s and 6.3 kbit/s. This
Recommendation specifies a coded representation that can be used to compress the speech signal
component of multi-media services at a very low bit rate. The audio is encoded in 30 ms frames.
A G.723.1 frame can be one of three sizes: 24 octets (6.3 kb/s frame), 20 octets (5.3 kb/s frame), or 4 octets.
These 4-octet frames are called SID frames (Silence Insertion Descriptor) and are used to specify comfort
noise parameters.
Table 190: G.723.1 Features
Feature Description

Voice Activity Detection (VAD) Supports the annex A, which is the built-in support of VAD in G.723.1.
Payload type 4
Available for fax No
Available for modem No
G.726
Standards Supported • ITU-T Recommendation G.726: 40, 32, 24, 16 kbit/s adaptive
differential pulse code modulation (ADPCM)
Algorithm recommended for conversion of a single 64 kbit/s A-law or U-law PCM channel encoded at 8000
samples/s to and from a 40, 32, 24, or 16 kbit/s channel. The conversion is applied to the PCM stream using
an Adaptive Differential Pulse Code Modulation (ADPCM) transcoding technique.
Feature Description
Packetization time Range of 10 ms to 30 ms with increments of 10 ms. See “G.726 Codecs

Voice Activity Detection (VAD) Two levels of detection are available: transparent or conservative. See
“Generic Voice Activity Detection (VAD)” on page 237 for more details.

Codec Descriptions Software Configuration Guide
Table 191: G.726 Features (Continued)
Feature Description
Comfort noise Uses custom comfort noise as defined in RFC 3389.

Payload type Configurable as per “G.726 Codecs Parameters” on page 240.
Available for fax Yes (32 kbps and 40 kbps)
Available for modem Yes (32 kbps and 40 kbps)
G.729
Standards Supported • ITU-T Recommendation G.729
Coding of speech at 8 kbit/s using conjugate structure-algebraic code excited linear prediction (CS-ACELP).
For all data rates, the sampling frequency (and RTP timestamp clock rate) is 8000 Hz.
A voice activity detector (VAD) and comfort noise generator (CNG) algorithm in Annex B of G.729 is
recommended for digital simultaneous voice and data applications; they can be used in conjunction with G.729
or G.729 Annex A. A G.729 or G.729 Annex A frame contains 10 octets, while the G.729 Annex B comfort
noise frame occupies 2 octets.
The Aastra unit supports G.729A and G.729AB for encoding and G.729, G.729A and G.729AB for decoding.
Feature Description

For reception, the range is extended from 10 ms to 100 ms with
Voice Activity Detection (VAD) Supports the annex B, which is the built-in support of VAD in G.729. See
“G.729 Codec Parameters” on page 242 for more details.
Payload type 18
Available for fax No

Chapter 28 - Voice & Fax Codecs Configuration Codec Descriptions
Clear Mode
Standards Supported • RFC 4040: RTP Payload Format for a 64 kbit/s Transparent
Call
The Clear Mode codec is similar to the G.711 codec but without any modification of the 64 kbit/s payload (no
encoding or decoding). The Clear Mode codec thus does not have echo cancellation and a fix jitter buffer.
Clear Mode is a method to carry 64 kbit/s channel data transparently in RTP packets. This codec always uses
the RTP transport.
Table 193: Clear Mode Features
Feature Description
Packetization time Range of 10 ms to 30 ms with increments of 10 ms. See “Clear Mode

Codec Parameters” on page 243 for more details.
Voice Activity Detection (VAD) N/A
Comfort noise N/A
Payload type Configurable as per “Clear Mode Codec Parameters” on page 243.
Clear Channel
Call
The Clear Channel codec is similar to the G.711 codec but without any modification of the 64 kbit/s payload
(no encoding or decoding). The Clear Channel codec thus does not have echo cancellation and a fix jitter
buffer. Clear Channel is a method to carry 64 kbit/s channel data transparently in RTP packets. The Clear
Channel codec follows the specification of RFC 4040 and uses the “X-CLEAR-CHANNEL” mime type instead
of the “CLEARMODE” mime type.
This codec always uses the RTP transport.
Table 194: Clear Channel Features
Feature Description
Packetization time Range of 10 ms to 30 ms with increments of 10 ms. See “Clear Channel

Codec Parameters” on page 245 for more details.
Comfort noise N/A
Payload type Configurable as per “Clear Channel Codec Parameters” on page 245.

Codec Descriptions Software Configuration Guide
X-CCD Clear Channel

Call
The Clear Channel codec is similar to the G.711 codec but without any modification of the 64 kbit/s payload
(no encoding or decoding). The X-CCD Clear Channel codec thus does not have echo cancellation and a fix
jitter buffer. The X-CCD Clear Channel is a method to carry 64 kbit/s channel data transparently in RTP
packets. The Clear Channel codec follows the specification of RFC 4040 and uses the “X-CCD” mime type
instead of the “CLEARMODE” mime type.
This codec always uses the RTP transport.
Table 195: X-CCD Clear Channel Features
Feature Description
Packetization time Range of 10 ms to 100 ms with increments of 1 ms. See “X-CCD Clear
Channel Codec Parameters” on page 246 for more details.
Comfort noise N/A
Payload type Configurable as per “X-CCD Clear Channel Codec Parameters” on
page 246.
T.38
Standards Supported • ITU-T Recommendation T.38 version 0
T.38 fax relay is a real-time fax transmission; that is, two fax machines communicating with each other as if
there were a direct phone line between the two. T.38 is called a fax relay, which means that instead of sending
inband fax signals, which implies a loss of signal quality, it sends those fax signals out-of-band in a T.38
payload, so that the remote end can reproduce the signal locally.
Table 196: T.38 Features
Feature Description
Packetization time N/A

Payload type N/A
Available for voice No
T.38 is an unsecure protocol, thus will not be used along with secure RTP (SRTP), unless the Allow Unsecure
T.38 with Secure RTP parameter has been set to Enable. See “Chapter 32 - Security” on page 377 for more
details.

Chapter 28 - Voice & Fax Codecs Configuration Codec Parameters
Codec Parameters
The Codec section allows you to enable or disable the codecs of the Aastra unit, as well as access the codec-
specific parameters.
Standards Supported • draft-choudhuri-sip-info-digit-00

• ITU-T Recommendation Q.24: Multifrequency push-button
signal reception
• RFC 2833: RTP Payload for DTMF Digits, Telephony Tones
and Telephony Signals
• RFC 1890: RTP Profile for Audio and Video Conferences with
Minimal Control

 Specific configurations that override the default configurations. You can define specific
configurations for each endpoint in your Aastra unit.
 To enable or disable the codecs:

1. In the web interface, click the Telephony link, then the CODECS sub-link.
Figure 106: Telephony – Codecs Web Page
3 4 5 6
2. Select to which endpoint (interface) you want to apply the changes in the Select Endpoint drop-
down menu at the top of the window.
You have the choice between Default and the interfaces of your Aastra unit. The number of
interfaces available vary depending on the Aastra unit model you have.
You can also perform this operation in the codec-specific pages.
3. Select whether or not you want to override one or more of the available default codecs parameters
in the Endpoint Specific column of the corresponding codec(s).
This column is available only in the specific endpoints configuration.
4. Enable one or more codecs for voice transmission by selecting Enable in the Voice column of the
corresponding codec(s).
This indicates if the codec can be selected for voice transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.

Generic Voice Activity Detection (VAD) Software Configuration Guide

5. Enable one or more codecs for data transmission by selecting Enable in the Data column of the
corresponding codec(s).
This indicates if the codec can be selected for data transmission. If enabled, this codec is listed as
6. Click the button to access the corresponding codec-specific parameters.
These parameters are described in the following sections.
Generic Voice Activity Detection (VAD)
VAD defines how the Aastra unit sends information pertaining to silence. This allows the unit to detect when
the user talks, thus avoiding to send silent RTP packets. This saves on network resources. However, VAD
may affect packets that are not really silent (for instance, cut sounds that are too low). VAD can thus slightly
affect the voice quality.
 To set the generic Voice Activity Detection (VAD)

1. In the Generic Voice Activity Detection (VAD) section, select whether or not you want to override
the VAD parameters set in the Default configuration in the Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
Figure 107: Generic Voice Activity Detection (VAD) Section
1
2
2. Enable the G.711 and G.726 Voice Activity Detection (VAD) by selecting the proper setting in the
Enable (G711 and G726) drop-down menu.
Table 197: G.711/G.726 VAD Settings
Setting Description
Disable VAD is not used.

Transparent VAD is enabled. It has low sensitivity to silence periods.
Conservative VAD is enabled. It has normal sensitivity to silence periods.
The difference between transparent and conservative is how “aggressive” the algorithm considers
something as an inactive voice and how “fast” it stops the voice stream. A setting of conservative is
a little bit more aggressive to react to silence compared to a setting of transparent.
G.711 Codec Parameters
The following are the G.711 codec parameters you can set. There are two sections for G.711:
 G.711 a-law
 G.711 u-law
These sections use the same parameters, so only one of them is described below.

Chapter 28 - Voice & Fax Codecs Configuration G.711 Codec Parameters
 To set the G.711 codec parameters:

1. In the CODEC section of the CODECS page, click the button at the right of the corresponding
G.711 codec to access the codec-specific parameters.
Figure 108: G.711 a-law Section
3
4
5
6
7
8
3. Select whether or not you want to override the G.711 parameters set in the Default configuration in
the Use Endpoint Specific drop-down menu.
You can also perform this operation in the main CODEC section.
4. Enable the G.711 codec for voice transmission by selecting Enable in the Voice Transmission drop-
down menu.
5. Set the default priority for voice in the Voice Priority field.
This sets the priority between different codecs. Codecs with a higher priority are used first, a priority
of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority
2. The maximum priority is 10.
The Aastra unit uses an internal order for codecs with the same priority.
Note: The codec used is also related to the SIP negotiation. The priority order affects the SIP negotiation,
which decides on the codec to use.
6. Enable the G.711 codec for data transmission by selecting Enable in the Data Transmission drop-
down menu.
7. Set the default priority for data in the Data Priority field.

G.723 Codec Parameters Software Configuration Guide
8. Select the minimum and maximum packetization time values for the codec in the Minimum
Packetization Time and Maximum Packetization Time drop-down menus.
The packetization time (also called packetization period or ptime) is the duration, in ms, of the voice
packet. The range is from 10 ms to 30 ms with increments of 10 ms.
For the reception, the range is extended from 10 ms to 100 ms with increments of 1 ms only if the
stream is not encrypted (SRTP).
You can also access the specific parameters of another codec by selecting the codec in the Select
CODEC drop-down menu at the top of the page.
The following are the G.723 codec parameters you can set.
Note that the G.723 codec is not available on the Asatra TA7102i Series models.

1. In the CODEC section of the CODECS page, click the button at the right of the G.723 codec
to access the codec-specific parameters.
Figure 109: G.723 Section
3
4
5
6
7
down menu.

Chapter 28 - Voice & Fax Codecs Configuration G.726 Codecs Parameters
6. Select the G.723 bit rate in the Bit Rate drop-down menu.
You have the following choices:
• 53 Kbs
• 63 Kbs
kstream is not encrypted (SRTP).
G.726 Codecs Parameters
The following are the G.726 codecs parameters you can set. There are four sections for G.726:
 G.726 16 Kbps
 G.726 24 Kbps
 G.726 32 Kbps
 G.726 40 Kbps
These sections offer almost the same parameters, except that you cannot use the G.726 16 Kbps and
G.726 24 Kbps codecs for fax transmission.
 To set the G.726 codecs parameters:


G.726 Codecs Parameters Software Configuration Guide
3
4
5
6
7
8
9
4. Enable the corresponding G.726 codec for voice transmission by selecting Enable in the Voice
Transmission drop-down menu.
6. Enable the codec for data transmission by selecting Enable in the Data Transmission drop-down
menu.
This menu is not available for the G.726 16 Kbps and G.726 24 Kbps codecs.
This field is not available for the G.726 16 Kbps and G.726 24 Kbps codecs.
8. Set the G.726 actual RTP dynamic payload type used in an initial offer in the Payload Type field.

Chapter 28 - Voice & Fax Codecs Configuration G.729 Codec Parameters
The payload types available are as per RFC 3551. The values range from 96 to 127. The default
values are as follows:
Table 198: G.726 Default Payload Type
Codec Default Value
G.726 (16 kbps) 97

G.726 (24 kbps) 98
G.726 (32 kbps) 99
G.726 (40 kbps) 100
9. Select the minimum and maximum packetization time values for the G.726 codec in the Minimum
The following are the G.729 codec parameters you can set.

1. In the CODEC section of the CODECS page, click the button at the right of the G.729 codec
3
4
5
6
7
3. In the G.729 section, select whether or not you want to override the G.729 parameters set in the
Default configuration in the Use Endpoint Specific drop-down menu.

Clear Mode Codec Parameters Software Configuration Guide
down menu.
For reception, the range is extended from 10 ms to 100 ms with increments of 10 ms only if the
7. Select the G.729 Voice Activity Detection (VAD) in the Built-in Voice Activity Detection (VAD) drop-
down menu.
Table 199: G.729 VAD
Disable G.729 uses annex A only.

Enable G.729 annex A is used with annex B. Speech frames are only sent during talkspurts
(periods of audio activity). During silence periods, no speech frames are sent, but
Comfort Noise (CN) packets containing information about background noise may
be sent in accordance with annex B of G.729.
VAD defines how the Aastra unit sends information pertaining to silence. This allows the unit to
detect when the user talks, thus avoiding to send silent RTP packets. This saves on network
resources. However, VAD may affect packets that are not really silent (for instance, cut sounds that
are too low). VAD can thus slightly affect the voice quality.
G.729 has a built-in VAD in its Annex B version. It is recommended for digital simultaneous voice
and data applications and can be used in conjunction with G.729 or G.729 Annex A. A G.729 or
G.729 Annex A frame contains 10 octets, while the G.729 Annex B frame occupies 2 octets. The
CN packets are sent in accordance with annex B of G.729.
Clear Mode Codec Parameters
The following are the Clear Mode codec parameters you can set.

Chapter 28 - Voice & Fax Codecs Configuration Clear Mode Codec Parameters
 To set the Clear Mode codec parameters:

1. In the CODEC section of the CODECS page, click the button at the right of the Clear Mode
codec to access the codec-specific parameters.
Figure 112: Clear Mode Section
3
4
5
6
7
8
9
3. Select whether or not you want to override the Clear Mode parameters set in the Default
configuration in the Use Endpoint Specific drop-down menu.
4. Enable the Clear Mode codec for voice transmission by selecting Enable in the Voice Transmission
drop-down menu.
6. Enable the Clear Mode codec for data transmission by selecting Enable in the Data Transmission
drop-down menu.
8. Set the Clear Mode RTP dynamic payload type used in an initial offer in the Payload Type field.

Clear Channel Codec Parameters Software Configuration Guide
value is 125.
Clear Channel Codec Parameters
The following are the Clear Channel codec parameters you can set.
 To set the Clear Channel codec parameters:

1. In the CODEC section of the CODECS page, click the button at the right of the Clear Channel
codec to access the codec-specific parameters.
Figure 113: Clear Channel Section
3
4
5
6
7
8
9
3. Select whether or not you want to override the Clear Channel parameters set in the Default
4. Enable the Clear Channel codec for voice transmission by selecting Enable in the Voice

Chapter 28 - Voice & Fax Codecs Configuration X-CCD Clear Channel Codec Parameters
6. Enable the Clear Channel codec for data transmission by selecting Enable in the Data
8. Set the Clear Channel RTP dynamic payload type used in an initial offer in the Payload Type field.
value is 125.
X-CCD Clear Channel Codec Parameters
The following are the X-CCD Clear Channel codec parameters you can set.
 To set the Clear Channel codec parameters:

1. In the CODEC section of the CODECS page, click the button at the right of the X CCD codec

X-CCD Clear Channel Codec Parameters Software Configuration Guide
Figure 114: X CCD Section
3
4
5
6
7
8
9
3. Select whether or not you want to override the X CCD parameters set in the Default configuration
in the Endpoint Specific drop-down menu.
4. Enable the X CCD codec for voice transmission by selecting Enable in the Voice Transmission
drop-down menu.
6. Enable the X CCD codec for data transmission by selecting Enable in the Data Transmission drop-
down menu.
8. Set the X CCD RTP dynamic payload type used in an initial offer in the Payload Type field.
value is 125.

Chapter 28 - Voice & Fax Codecs Configuration Fax Parameters
Fax Parameters
The Aastra unit handles G3 fax transmissions at speeds up to 14.4 kbps. Automatic fax mode detection is
standard on all endpoints. Real-Time Fax Over UDP with the T.38 protocol stack is also available.
A fax call works much like a regular voice call, with the following differences:
1. The fax codec may be re-negotiated by using a re-INVITE.
2. The goal of the re-INVITE is to allow both user agents to agree on a fax codec, which is either:
a. Clear channel (G.711 or G.726) without Echo Cancellation nor Silence Suppression
(automatically disabled).
b. T.38.
3. Upon fax termination, if the call is not BYE, the previous voice codec is recovered with another re-
INVITE.
All endpoints of the Aastra unit can simultaneously use the same codec (for instance, T.38), or a mix of any
of the supported codecs. Set and enable these codecs for each endpoint.
Clear Channel Fax

The Aastra unit can send faxes in clear channel. The following is a clear channel fax call flow:
Figure 115: Clear Channel Fax Call Flow
INVITE
[…] […]
m=audio 5004 RTP/AVP 18 0 13 m=audio 5006 RTP/AVP 18 0 13
a=rtpmap:18 G729/8000 a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000 a=rtpmap:0 PCMU/8000
200 OK
Ringing/Trying
ACK
RTP=G.729 (Voice Call)
Fax Tone Detected
User User
Agent RTP=PCMU (Echo Cancellation + Silence Suppression = disabled)
Agent
#1 #2
No re-INVITE!!
 There is no need for a re-INVITE since the far end already supports the
data codec (PCMU).
 When your SDP capabilities are inserted in a SIP packet, it implies that
you can receive any of these capabilities at any given time without notice.
 In this case, both ends should switch to clear channel automatically upon
detection of the fax transmission.
Fax is terminated
BYE
200 OK

Fax Parameters Software Configuration Guide
DSP Limitation
The Aastra unit currently suffers from a limitation of its DSP. Because of this limitation, the voice does not
switch back to the original negotiated codec after a clear channel fax is performed.
The Aastra unit cannot detect the end of a clear channel fax, which means that the unit cannot switch back to
the original negotiated codec if this codec was not a clear channel codec, e.g., a session established in G.729.
When the unit detects a fax, it automatically switches to a negotiated clear channel codec such as PCMU (if
there is no T.38 or if T.38 negotiation failed). Once the fax is terminated, the Aastra unit is not notified by the
DSP. The unit thus stays in the clear channel codec and does not switch back to G.729.
T.38 Fax
The Aastra unit can send faxes in T.38 mode over UDP. T.38 is used for fax if both units are T.38 capable;
otherwise, transmission in clear channel over G.711 as defined is used (if G.711 µ-law and/or G.711 A-law are
enabled). If no clear channel codecs are enabled and the other endpoint is not T.38 capable, the fax
transmission fails.
Caution: The Aastra unit opens the T.38 channel only after receiving the “200 OK” message from the peer.
This means that the Aastra unit cannot receive T.38 packets before receiving the “200 OK”. Based on RFC
3264, the T.38 channel should be opened as soon as the unit sends the “INVITE” message.
The quality of T.38 fax transmissions depends upon the system configuration, type of call control system used,
type of Aastra units deployed, as well as the model of fax machines used. Should some of these conditions
be unsatisfactory, performance of T.38 fax transmissions may vary and be reduced below expectations.
Note: Aastra recommends not to use a fax that does not send a CNG tone. If you use such a fax to send a
fax communication to the public network, this might result in a communication failure.
The following is a T.38 fax call flow:
Figure 116: T.38 Fax Call Flow
INVITE
[…] […]
m=audio 5004 RTP/AVP 0 18 4 8 13 m=audio 5006 RTP/AVP 0 18 4 8 13
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
200 OK
Ringing/Trying
ACK
Fax Tone Detected

INVITE
[…] […]
m=image 6006 udptl t38 m=image 6006 udptl t38
a=T38MaxBitRate:14400 a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF a=T38FaxRateManagement:transferredTCF
User a=T38FaxFillBitRemoval:0 a=T38FaxFillBitRemoval:0 User
a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingMMR:0
Agent a=T38FaxTranscodingJBIG:0 a=T38FaxTranscodingJBIG:0 Agent
#1 a=T38FaxVersion:0
a=T38FaxUdpEC:t38UDPRedundancy
a=T38FaxVersion:0 #2
200 OK a=T38FaxUdpEC:t38UDPRedundancy
Trying
ACK
Fax is terminated
INVITE
[…] […]
m=audio 5004 RTP/AVP 0 18 4 8 13 m=audio 5006 RTP/AVP 0 18 4 8 13
a=rtpmap:8 PCMA/8000 a=rtpmap:8 PCMA/8000
200 OK
Trying
ACK
BYE
200 OK

T.38 Parameters Configuration

The following are the T.38 codec parameters you can set.
 To set the T.38 codec parameters:

Figure 117: T.38 Section
4
5
6
7
8
9
10
3. In the T.38 section, select whether or not you want to override the T.38 parameters set in the Default
4. Enable the T.38 codec by selecting Enable in the Enable drop-down menu.
5. Set the default priority for fax in the Priority field.
Note: Currently, the only T.38 priority accepted is 10. Priority between 1 and 9 is refused.
6. Set the number of redundancy packets sent with the current packet in the Redundancy Level field.
This is the standard redundancy offered by T.38. Available values range from 1 to 5. Please see
step 7 for additional reliability options for T.38.
7. Set the T.38 input signal detection threshold in the Detection Threshold drop-down menu.
Lowering the threshold allows detecting lower amplitude fax signals. The following values are
available:
• Default: (-26 dB)
• Low: (-31 dB)
• Lowest: (-43 dB)

Fax Parameters Software Configuration Guide
8. For additional reliability, define the number of times T.38 packets are retransmitted in the Frame
Redundancy Level field.
This field is available only in the default endpoint configuration.
This only applies to the T.38 packets where the PrimaryUDPTL contains the following T.38 data
type:
• HDLC_SIG_END,
• HDLC_FCS_OK_SIG_END,
• HDLC_FCS_BAD_SIG_END and
• T4_NON_ECM_SIG_END
9. Define whether or not the Aastra unit sends no-signal packets during a T.38 fax transmission in the
No Signal drop-down menu.
This menu is available only in the default endpoint configuration.
When enabled, the unit ensures that, during a T.38 fax transmission, data is sent out at least every
time the No Signal Timeout delay expires. The Aastra unit sends no-signal packets if no meaningful
data have been sent for a user-specified period of time.
10. Set the period, in seconds, at which no-signal packets are sent during a T.38 transmission in the No
Signal Timeout field.
This field is available only in the default endpoint configuration.
No-signal packets are sent out if there are no valid data to send.
Data Codec Selection Procedure

The Aastra unit follows a procedure when selecting data codec. This procedure is the default behaviour of the
Aastra unit. Some interop variables may modify this procedure. Tones are detected on the analog ports only.

Figure 118: Data Codec Selection Procedure
Voice /
Voiceband
data call
Complete fax/modem codec
selection procedure .
Tones are detected on the telephony ports except for
the CED, which can also be detected on the IP side CNG tone No CED tone V.21 tone
No No Continue call
detected? detected? detected?
Yes
Yes
Evaluate Yes
BehaviorOnCed
ToneDetection
Passthrough
Stop sending Faxmode
voice codec. Start T.38 is
Yes
buffering T.38 enabled?
packets
Current voice
Send a re-INVITE Continue call with
No codec is data - Yes
for T.38 current codec
capable?
No
Evaluate
BehaviorOnT38Invite List of
T.38 accepted negotiated Switch to highest
No NotAccepted.Behavior
by peer? codecs negotiated priority
according to error Yes
code contains a data codec and
data-capable continue call
Yes codec?
ReInviteForClear
ChannelOnly (default) No
Start sending T.38
packets Switch to highest
configured priority G.711
data codec
At least one data (PCMU if all disabled )
codec is enabled
on the device ? ReInviteOn
Continue T.38 fax NoNegotiated
DataCodec
No Evaluate
ReEstablish interopB ehavior ReInviteOn
Yes Audio Continue call
DropCall OnMachine FaxT38Only
Detection
UsePrevious
MediaNegiciation
Send SIP re- Disable fax/modem detection
INVITE for data- and send SIP re-INVITE for
capable codecs voice -capable codecs
New codec
Terminate call No accepted by
peer ?
Yes
Switch to
Restore the previouly
selected data
used voice codec and
codec and
continue call
continue call

C H A P T E R
29 Security
This chapter describes how to properly configure the security parameters of the Aastra unit.
Standards Supported • RFC 3711: The Secure Real-time Transport Protocol (SRTP)
(Supports only the AES-CM encryption)
• RFC 3830: MIKEY: Multimedia Internet KEYing (Compliant
for method Pre-Shared Key only)
• RFC 4567: Key Management Extensions for Session
Description Protocol (SDP) and Real Time Streaming
Protocol (RTSP)
• RFC 4568: SDES: Security Descriptions for Media Streams
Introduction
You can define security features on the Aastra unit. This section applies to media security parameters.
Applying security on the Aastra unit involves several steps:
 Properly set the time on the Aastra unit by configuring a valid SNTP server (“SNTP
Configuration” on page 93) and time zone (“Time Configuration” on page 94).
 Transfer a valid CA certificate into the Aastra unit (“Chapter 46 - Certificates Management” on
page 557).
 Use secure signalling by enabling the TLS transport protocol (“Chapter 28 - SIP Transport
Parameters” on page 305).
Caution: If you enable Secure RTP (SRTP) on at least one line, it is acceptable to have the secure SIP
transport (TLS) disabled for testing purposes. However, you must never use this configuration in a
production environment, since an attacker could easily break it. Enabling TLS for SIP Transport is strongly
recommended and is usually mandatory for security interoperability with third-party equipments.
Caution: When using a codec other than G.711, enabling Secure RTP (SRTP) has an impact on the Aastra
unit’s overall performance as SRTP requires CPU power. The more lines use SRTP, the more overall
performance is affected. See also “DSP Limitation” on page 429 for more details on resources limitations
with SRTP and conferences.
 Use secure media by:

• Defining the SRTP/ SRTCP base port (“Base Ports Configuration” on page 397).
• Setting the RTP secure mode to “Secure” or “Secure with fallback” (this section).
Security Parameters
The Security section allows you to secure the RTP stream (media) of the Aastra unit.
Since the SRTP encryption and authentication needs more processing, the number of calls that the Aastra unit
can handle simultaneously may be reduced, depending of the codecs enabled. You could set the Aastra unit
not to impact the number of simultaneous calls by enabling only G.711 codecs and disabling every other voice
or data codec, even T.38.

Chapter 29 - Security Security Parameters
The Aastra unit supports the MIKEY protocol using pre-shared keys (MIKEY-PS) or the SDES protocol for
negotiating SRTP keys.
 To set the RTP stream security parameters:

1. In the web interface, click the Media link, then the Security sub-link.
Figure 119: Media – Security Web Page
4
5
6
3. Select whether or not you want to override one or more of the available default security parameters
in the Endpoint Specific drop-down menu.
4. In the Security section of the Security page, select the RTP payload mode in the Mode drop-down
menu.
The unit relies on these modes when negotiating an audio stream.
Table 200: Default RTP Mode
Mode Description
Unsecure The Aastra unit supports only unsecure RTP. It rejects secure RTP
offers it receives.
Secure The Aastra unit supports only secure RTP. It rejects unsecure RTP
offers it receives.
Secure with fallback The Aastra unit supports both secure and unsecure RTP. It prioritizes
secure RTP but permits unsecure RTP fallback when the remote peer
does not support security.
The TLS SIP transport must usually be enabled for secure audio negotiation via SDP (refer to the
Caution box above). See “Chapter 28 - SIP Transport Parameters” on page 305 for more details.
The RTP mode is reflected in the SIP/SDP payload, with a RTP/AVP for unsecure RTP, and a RTP/
SAVP for secure RTP.
The following basic rules apply when sending units capabilities via SDP:
• When the RTP mode is set to Unsecure, the Aastra unit offers/answers with only one
active RTP/AVP audio stream. Any other audio stream present in the offer is disabled
in the answer.
• When the RTP mode is set to Secure, the Aastra unit offers/answers with only one
active RTP/SAVP audio stream. Any other audio stream present in the offer is disabled
in the answer.

Security Parameters Software Configuration Guide
• When the RTP mode is set to Secure with fallback, the Aastra unit offers one RTP/AVP
and one RTP/SAVP audio streams. The unit answers with only the most secure stream.
• If the remote unit answers to an offer with both RTP/AVP and RTP/SAVP streams
enabled, a new offer is sent with only RTP/SAVP enabled.
5. Select the key management protocol for SRTP in the Key Management drop-down menu.
Table 201: Key Management Protocol
Mikey Use MIKEY (Multimedia Internet KEYing).

Sdes Use SDES (Security DEScriptions).
This parameter has no effect if the Mode parameter is set to Unsecure.

If the unit receives an offer with both MIKEY and SDES, only the configured key management
protocol is kept.
6. Select the encryption type to be used with SRTP in the Encryption drop-down menu.
Encryption Description
Null No encryption. It is ignored for the Sdes Key Management as

defined in Step 3. Use only for debug.
AesCm128 AES (Advanced Encryption Standard) Counter Mode 128 bits.
This parameter has no effect if the Mode parameter is set to Unsecure.

7. Select whether or not to enable T.38 even if the call has been established previously in SRTP in the
Allow Unsecure T.38 with Secure RTP drop-down menu.
Mode Description
Disable T.38 is disabled for SRTP calls.

Enable T.38 is enabled for SRTP calls.
Caution: Enabling this parameter opens a security hole,
because T.38 is an unsecure protocol.
This menu is available only in the default configuration.

Note that this parameter has no effect if the Mode parameter is set to Unsecure.
Enforcing Symmetric RTP

For each bi-directional RTP streams, you can define whether or not to enforce that incoming RTP packets are
from the same source as the destination of outgoing RTP packets.
Enforcing symmetric RTP may prevent legitimate RTP streams coming from a media server from being
processed, for example: Music and conferencing servers.

Chapter 29 - Security Security Parameters

Table 204: Enforce Symmetric RTP Parameters
disable Accept packets from all sources. This is the default value.
enable Silently discard incoming RTP packets with source address and port differing from the
destination address and port of outgoing packets.
 To enforce symmetric RTP:

1. In the miptMIB, set the enforceSymmetricRtpEnable variable with the proper behaviour.
mipt.enforceSymmetricRtpEnable="Value"
Figure 120: Symmetric RTP Values
Value Meaning
0 disable
1 enable

C H A P T E R
30 RTP Statistics Configuration
The Aastra unit collects meaningful statistics that can be read via the web interface. This chapter describes
how to read and configure the RTP statistics.
Note that the RTP statistics are also available via SNMP and CLI.
Statistics Displayed
The Aastra unit collects two types of statistics:

 statistics for the last 10 connections
 statistics for the last 10 collection periods
The Connection Statistics section displays the statistics for the last 10 connections. You can use the Display
All button to display more information or the Display Overview button to display less information.
The Connection Period Statistics section displays the statistics for the last 10 periods. The period duration is
defined in the Statistics Configuration section. You can use the Display All button to display more information
or the Display Overview button to display less information.
Figure 121: Telephony – RTP Stats Web Page
The following table describes the statistics available.

Table 205: Statistics Displayed
Statistic Connection Statistics Collection Period Statistics
Octets Tx Number of octets transmitted during the Number of octets transmitted during the
connection. collection period. This value is obtained by
cumulating the octets transmitted in all
connections that were active during the
collection period.

Chapter 30 - RTP Statistics Configuration Statistics Displayed
Table 205: Statistics Displayed (Continued)
Octets Rx Number of octets received during the Number of octets received during the
cumulating the octets received in all
collection period.
Packets Tx Number of packets transmitted during the Number of packets transmitted during the
cumulating the packets transmitted in all
collection period.
Packets Rx Number of packets received during the Number of packets received during the
cumulating the packets received in all
collection period.
Packets Lost Number of packets lost during the Number of packets lost during the collection
connection. This value is obtained by period. This value is obtained by cumulating
substracting the expected number of the packets lost in all connections that were
packets based on the sequence number active during the collection period.
from the number of packets received.
Min. Jitter Minimum interarrival time, in ms, during the Minimum interarrival time, in ms, during the
connection. All RTP packets belonging to collection period. This value is the lowest
the connection and received at the RTP interarrival jitter for all connections that were
level are considered in the calculation. active during the collection period.
Max. Jitter Maximum interarrival time, in ms, during the Maximum interarrival time, in ms, during the
connection. All RTP packets belonging to collection period. This value is the highest
the connection and received at the RTP interarrival jitter for all connections that were
level are considered in the calculation. active during the collection period.
Avg. Jitter Average interarrival time, in ms, during the Average interarrival time, in ms, during the
connection. All RTP packets belonging to collection period. This value is the weighted
the connection and received at the RTP average of the interarrival jitter for all
level are considered in the calculation. connections that were active during the
collection period. For each connection, the
total jitter of packets received during the
collection period and the total number of
packets received during the collection period
are used in the weighted average
calculation.
Min. Latency Minimum latency, in ms, during the Minimum latency, in ms, during the
connection. The latency value is computed collection period. This value is the lowest
as one half of the round-trip time, as latency for all connections that were active
measured through RTCP. during the collection period.
Max. Latency Maximum latency, in ms, during the Maximum latency, in ms, during the
connection. The latency value is computed collection period. This value is the highest
as one half of the round-trip time, as latency for all connections that were active
measured through RTCP. during the collection period.

Statistics Configuration Software Configuration Guide
Table 205: Statistics Displayed (Continued)
Avg. Latency Average latency, in ms, during the Average latency, in ms, during the collection
connection. The latency value is computed period. This value is the weighted average
as one half of the round-trip time, as of the latency for all connections that were
measured through RTCP. active during the collection period. For each
connection, the total latency of packets
received during the collection period and the
total number of packets received during the
collection period are used in the weighted
average calculation.
Statistics Configuration
You can define how to collect the statistics. The statistics are sent as syslog messages, so you must properly
set the syslog information before setting the statistics. You must set the Media IP Transport (MIPT) service to
the Info or Debug level. See “Syslog Daemon Configuration” on page 71 for more details on how to configure
the Syslog.
 To configure how to collect statistics:

1. In the web interface, click the Telephony link, then the RTP Stats sub-link.
Figure 122: Telephony – RTP stats Web Page
2
3
4
2. Set the Collection Period field with the collection period duration in minutes.
Putting a value of 0 disables the collection period statistics feature.
3. Set the End-of-Connection Notification drop-down menu with the proper behaviour.
Table 206: End-of-Connection Notification
Enable Notifications are generated.

Disable Notifications are not generated.

Chapter 30 - RTP Statistics Configuration Channel Statistics
4. Set the End-of-Period Notification drop-down menu with the proper behaviour.
Table 207: End-of-Period Notification
Enable Notifications are generated.

Disable Notifications are not generated.

• To save your settings, click Submit.
• To save your settings and reset the statistics of the current period., click Submit &
Reset Current Collection Period Statistics.
The previous periods are left unchanged.
Channel Statistics
This section describes how to access data available only in the MIB parameters of the Aastra unit. You can
display these parameters as follows:
The channel statistics are cumulated RTP statistics for all calls using a specific channel of a telephony
interface. Statistics are updated at the end of each call.
The statistics are associated to the channel in use at the end of the call. In some cases, such as in hold/resume
scenarios, the channel assignment may change during a call. This can result in discrepancies between the
RTP statistics and the actual usage of the telephony interface.
The following are the channel statistics the Aastra unit keeps.
Table 208: Channel Statistics
MIB Variable Statistics Description
PacketsSent Number of packets transmitted on the channel since

service start. This value is obtained by cumulating the
packets transmitted in all the connections that ended
during the collection period.
PacketsReceived Number of packets received on the channel since
packets received in all the connections that ended
BytesSent Number of bytes transmitted on the channel since
bytes transmitted in all the connections that ended
BytesReceived Number of bytes received on the channel since service
start. This value is obtained by cumulating the bytes
received in all the connections that ended during the
collection period.
AverageReceiveInterarr Average interarrival time, in microseconds, for the
ivalJitter channel since service start. This value is based on the
average interarrival jitter of each call ended during the
collection period. The value is weighted by the duration
of the calls.

Channel Statistics Software Configuration Guide
 To display channel statistics:

1. In the miptMIB, go to the ChannelStatistics table.
You can also use the following line in the CLI:
get mipt.channelStatistics
 To reset channel statistics values to zero:

1. In the miptMIB, set ChannelStatistics.Reset to Reset for the endpoint to reset.
set mipt.channelStatistics.Reset=Reset
2. In the miptMIB, set ChannelStatistics[EpChannelId=channelStatisticsEpChannelId].Reset

to Reset to reset only one specific endpoint.
where:
• channelStatisticsEpChannelId is the string that identifies the combination of an
endpoint and a channel. The endpoint name is the same as the EpId used to refer to
endpoints in other tables. On endpoints with multiple channels, the channel number
must be appended at the end of the endpoint name, separated with a dash.
set mipt.channelStatistics[EpChannelId=channelStatisticsEpChannelId].Reset=Reset
Examples:
Slot3/E1T1-12 refers to endpoint Slot3/E1T1, channel 12.
Phone-Fax1 refers to FXS endpoint Phone-Fax1 on a 4102s.
Port06 refers to FXS endpoint Port06 on 4108/4116/4124.
No channel number is appended to FXS endpoint strings because FXS lines do not support multiple
channels.

Chapter 30 - RTP Statistics Configuration Channel Statistics

C H A P T E R
31 Miscellaneous Media Parameters
This chapter describes how to configure parameters that apply to all codecs.
Standards Supported • draft-choudhuri-sip-info-digit-00.txt
 Jitter Buffer Configuration

 DTMF Transport Configuration
 Machine Detection Configuration
 Base Ports Configuration
Jitter Buffer Configuration
The Jitter Buffer section allows you to configure parameters to reduce jitter buffer issues.
 To set the jitter buffer parameters:

1. In the web interface, click the Media link, then the Misc sub-link.
Figure 123: Media – Misc Web Page
3
4
5
6
7
8
9
10
3. In the Jitter Buffer section, if you have selected a specific endpoint, select whether or not you want
to override the jitter buffer parameters set in the Default configuration in the Endpoint Specific drop-
down menu.
4. Select the jitter buffer level in the Level drop-down menu.

Chapter 31 - Miscellaneous Media Parameters Jitter Buffer Configuration
Jitter is an abrupt and unwanted variation of one or more signal characteristics, such as the interval
between successive pulses or the frequency or phase of successive cycles. An adaptive jitter buffer
usually consists of an elastic buffer in which the signal is temporarily stored and then retransmitted
at a rate based on the average rate of the incoming signal.
Table 209: Jitter Buffer Levels
Level Description
Optimize Latency The jitter buffer is set to the lowest effective value to minimize the latency.
Voice cut can be heard if the network is not optimal. The predefined values
are as follows:
• Minimum value: 10 ms
• Maximum value: 40 ms
Normal The jitter buffer tries to find a good compromise between the latency and the
voice quality. This setting is recommended in private networks. The
predefined values are as follows:
Optimize Quality The jitter buffer is set to a high value to minimize the voice cuts at the cost of
high latency. This setting is recommended in public networks. The
predefined values are as follows:
Fax / Modem The jitter buffer is set to maximum. The Fax/Modem transmission is very
sensitive to voice cuts but not to latency, so the fax has a better chance of
success with a high buffer. The predefined values are as follows:
Custom The jitter buffer uses the configuration of the Minimum and Maximum
variables (Steps 4 and 5).
5. If you have selected the Custom level, define the target jitter buffer length in the Minimum field of
the Voice Call part.
The adaptive jitter buffer attempts to hold packets to the minimal holding time. This is the minimal
delay the jitter buffer adds to the system. The minimal jitter buffer is in ms and must be equal to or
smaller than the maximal jitter buffer.
Values range from 0 ms to 135 ms. The default value is 30 ms. You can change values by
increments of 1 ms, but Aastra recommends to use multiples of 5 ms. The minimal jitter buffer
should be a multiple of ptime.
It is best not to set the minimal jitter value below the default value. Setting a minimal jitter buffer
below 5 ms could cause an error. Jitter buffer adaptation behaviour varies from one codec to
another. See “About Changing Jitter Buffer Values” on page 265 for more details.
6. If you have selected the Custom level, define the maximum jitter buffer length in the Maximum field
of the Voice Call part.
This is the highest delay the jitter buffer is allowed to introduce. The jitter buffer length is in ms and
must be equal to or greater than the minimum jitter buffer.
Values range from 0 ms to 135 ms. The default value is 125 ms. You can change values by
increments of 1 ms, but Aastra recommends to use multiples of 5 ms. The maximal jitter buffer
should be a multiple of ptime.
The maximum jitter buffer value should be equal to the minimum jitter buffer value + 4 times the
ptime value. Let’s say for instance that:
• Minimum jitter buffer value is 30 ms

Jitter Buffer Configuration Software Configuration Guide
• Ptime value is 20 ms
The maximum jitter buffer value should be: 30 + 4x20 = 110 ms
7. If you have selected the Custom level, define the voiceband data custom jitter buffer type in the
Playout Type drop-down menu of the Data Call part.
This is the algorithm to use for managing the jitter buffer during a call. The Nominal field value
serves as the delay at the beginning of the call and might be adapted afterwards based on the
selected algorithm.
Table 210: Voiceband Data Custom Jitter Buffer Type
Level Description
Adaptive The nominal delay varies based on the estimated packet jitter. Playout
Immediately adjustment is done immediately when the actual delay goes out of bounds of
a small window around the moving nominal delay.
Adaptive Silence The nominal delay varies based on the estimated packet jitter. Playout
adjustment is done based on the actual delay going out of bounds of a small
window around the moving nominal delay. The adjustment is deferred until
silence is detected (either from playout buffer underflow or by analysis of
packet content). Playout adjustment is also done when overflow or
underflow events occur.
Fixed The nominal delay is fixed to the value of the Nominal field value and does
not change thereafter. Playout adjustment is done when overflow or
underflow events occur.
8. If you have selected the Custom level, define the voiceband data jitter buffer minimal length (in
milliseconds) in the Minimum field of the Data Call part.
The voiceband data jitter buffer minimal length is the delay the jitter buffer tries to maintain. The
minimal jitter buffer MUST be equal to or smaller than the voiceband data maximal jitter buffer.
The minimal jitter buffer should be a multiple of ptime.
This value is not available when the Playout Type drop-down menu is set to Fixed.
9. If you have selected the Custom level, define the voiceband data custom jitter buffer nominal length
in the Nominal field of the Data Call part.
The jitter buffer nominal length (in milliseconds) is the delay the jitter buffer uses when a call begins.
The delay then varies depending on the type of jitter buffer.
In adaptive mode, the nominal jitter buffer should be equal to (voice band data minimal jitter buffer
+ voice band data maximal jitter buffer) / 2.
10. If you have selected the Custom level, define the default voiceband data custom jitter buffer
maximal length in the Maximum field of the Data Call part.
The jitter buffer maximal length (in milliseconds) is the highest delay the jitter buffer is allowed to
introduce. The maximal jitter buffer MUST be equal to or greater than the minimal jitter buffer.
The maximal jitter buffer should be a multiple of ptime.
The maximal jitter buffer should be equal to or greater than voiceband data minimal jitter buffer + (4
* ptime) in adaptive mode.
See “About Changing Jitter Buffer Values” on page 265 for more details.
About Changing Jitter Buffer Values

Aastra recommends to avoid changing the target and maximum jitter buffer values unless experiencing or
strongly expecting one of the following symptoms:
 If the voice is scattered, try to increase the maximum jitter buffer value.
 If the delay in the voice path (end to end) is too long, you can lower the target jitter value, but

Chapter 31 - Miscellaneous Media Parameters DTMF Transport Configuration
ONLY if the end-to-end delay measured matches the target jitter value.
For instance, if the target jitter value is 50 ms, the maximum jitter is 300 ms and the delay measured
is 260 ms, it would serve nothing to reduce the target jitter. However, if the target jitter value is
100 ms and the measured delay is between 100 ms and 110 ms, then you can lower the target jitter
from 100 ms to 30 ms.
Starting a Call in Voiceband Data Mode

You can define whether or not a call should be started in voiceband data mode.
Table 211: Voiceband Data Mode Parameters
Disable The call is started in voice mode. A fax/modem tone detection triggers a transition from voice
to voiceband data according to the configuration in the Machine Detection Group
(“Miscellaneous Media Parameters” on page 263).
Enable The call is started in voiceband data mode.
 To start a call in voiceband data mode:

1. In the telIfMIB, set the voiceband data mode in the InteropStartCallInVbdEnable variable.
telIf.InteropStartCallInVbdEnable="Value"
Table 212: Voiceband Data Mode Values
Value Method
0 Disable
1 Enable
DTMF Transport Configuration
The DTMF Transport section allows you to set the DTMF transport parameters of the Aastra unit.
 To set DTMF transport parameters:

2. In the DTMF Transport section of the Misc page, select whether or not you want to override the
DTMF transport parameters set in the Default configuration in the Endpoint Specific drop-down
menu.

DTMF Transport Configuration Software Configuration Guide
Figure 124: DTMF Transport Section
2
3
4
3. Select the DTMF transport type in the Transport Method drop-down menu.
The following choices are available:
Table 213: DTMF Transport Type Parameters
Transport Parameter Description
In-band The DTMFs are transmitted like the voice in the RTP
stream.
Out-of-band using RTP The DTMFs are transmitted as per RFC 2833. This
parameter also works with SRTP.
Out-of-band using SIP The DTMFs are transmitted as per draft-choudhuri-sip-info-
digit-00.
Signaling protocol Dependant The signalling protocol has the control to select the DTMF
transport mode. The SDP body includes both RFC 2833
and draft-choudhuri-sip-info-digit-00 in that order of
preference.
4. If you have selected the Out-of-band using SIP transport method, select the method used to
transport DTMFs out-of-band over the SIP protocol in the SIP Transport Method drop-down menu.
Table 214: DTMF Out-of-Band Transport Methods
Method Description
draftChoudhuriSipInfoDigit00 Transmits DTMFs by using the method defined in draft-

choudhuri-sip-info-digit-00. Only the unsolicited-digit part is
supported.
DTMF out-of-band
Certain compression codecs such as G.723.1 and G.729 effectively distort voice because they lose
information from the incoming voice stream during the compression and decompression phases. For
normal speech this is insignificant and becomes unimportant. In the case of pure tones (such as DTMF)
this distortion means the receiver may no longer recognize the tones. The solution is to send this
information as a separate packet to the other endpoint, which then plays the DTMF sequence back by re-
generating the true tones. Such a mechanism is known as out-of-band DTMF. The Aastra unit receives
and sends out-of-band DTMFs as per ITU Q.24. DTMFs supported are 0-9, A-D, *, #.

Table 214: DTMF Out-of-Band Transport Methods (Continued)
Method Description
Info DTMF Relay Transmits DTMFs by using a custom method. This custom
method requires no SDP negotiation and assumes that the other
peer uses the same method.
It uses a SIP INFO message with a content of type application/
dtmf-relay. The body of the message contains the DTMF
transmitted and the duration of the DTMF:
Signal= 1
Duration= 160
When transmitting, the duration is the one set in the
interopDtmfTransportDuration variable (see “DTMF
Transport over the SIP Protocol” on page 268).
When receiving, the duration of the DTMF received is not used
and the DTMF is played for 100 ms.
DTMFs are transmitted one at a time.
Available digits are “0123456789ABCD*#”. The Aastra unit also
supports the “,;p” characters when receiving DTMFs.
5. If you have selected the Out-of-band using RTP transport method, set the payload type in the
Payload Type field.
You can determine the actual RTP dynamic payload type used for the “telephone-event” in an initial
offer. The payload types available are as per RFC 1890. Available values range from 96 to 127.
DTMF Transport over the SIP Protocol

You can set the DTMF duration sent in the INFO message when using the Info DTMF Relay method to
transmit DTMFs (see “Miscellaneous Media Parameters” on page 263, Step 8 for more details).
 To set the DTMF duration sent in the INFO message:

1. In the sipEpMIB, set the DTMF duration sent in the INFO message when using the infoDtmfRelay
method to transmit DTMFs in the interopDtmfTransportDuration variable.
sipEp.interopDtmfTransportDuration="Value"
This value is expressed in milliseconds (ms). The default value is 100 ms.
DTMF Detection
The default DTMF detection parameters of the Aastra unit may sometimes not be enough to properly detect
the DTMFs. This section describes how to set additional DTMF detection parameters.

DTMF Transport Configuration Software Configuration Guide
DTMF Frequencies
The DTMF keypad is laid out in a 4x4 matrix, with each row representing a low frequency, and each column
representing a high frequency. For example, pressing a single key (such as '1') sends a sinusoidal tone of the
two frequencies (697 Hz and 1209 Hz). When the unit is configured to send DTMFs out-of-band, its DSP
detects these DTMFs, removes them from the RTP stream, and sends them out-of-band.
Table 215: DTMF Keypad Frequencies
Low/High (Hz) 1209 1336 1477 1633
697 1 2 3 A
770 4 5 6 B
852 7 8 9 C
941 * 0 # D
DTMF Detection Configuration

Below is a frequency spectrum analysis of a DTMF (9) with the Frequency in Hertz on the x axis and the Power
in dBm on the y axis. The low and high frequencies of the DTMF are in red and you can clearly see that they
are the most powerful frequencies in the signal.
Figure 125: DTMF Detection Example
 To configure the DTMF detection:

1. In the telIfMIB, define how the Rise Time criteria should be configured for DTMF detection in the
interopDtmfDetectionRiseTimeCriteria variable.
sipEp.interopDtmfDetectionRiseTimeCriteria="Value"


Table 216: DTMF Detection Values
Value Method
100 CheckSr Enables the Step Rise criteria and disables the Confirm DTMF SNR
criteria.
The Step Rise criteria compares the current frame energy to the high
frequency power of the previous frame. If the current frame energy is high
enough, then it passes the test, further validating the DTMF.
Disabling the Step Rise criteria may result in deteriorated talk-off
performance, but increases the detection of malformed DTMF.
200 ConfirmSnr Enable the Confirm DTMF SNR criteria and disable the Step Rise criteria.
The Confirm DTMF SNR criteria is an additional Signal-to-noise ratio test
performed before a confirmed DTMF report is sent to finally validate the
DTMF.
2. Set the interopDtmfDetectionPositiveTwist variable.

sipEp.interopDtmfDetectionPositiveTwist="Value"
When the high-group frequency of a DTMF is more powerful than the low-group frequency, the
difference between the high-group frequency absolute power and the low-group frequency absolute
power must be smaller than or equal to the value set in this variable. Otherwise, the DTMF is not
detected.
Raising this value increases the sensitivity of DTMF detection. Raising this value too high may also
cause false detections of DTMFs.
Using the Payload Type Found in the Answer

The default behaviour when sending an initial offer that contains an RFC 2833 payload type is to keep using
that payload type even if the response comes back with a different one. You can set the Aastra unit to rather
use the payload type found in the answer.
This feature is effective only if the Transport Method drop-down menu is set to Out-of-band using RTP (see
“Miscellaneous Media Parameters” on page 263 for more details).
Table 217: Payload Type in Answer
disable Keep using the initial payload type. This is the default value.
enable Use the RFC 2833 payload type found in the received answer.
 To use the payload type found in the answer:

1. In the sipEpMIB, set the interopUseDtmfPayloadTypeFoundInAnswer variable with the proper
behaviour.
sipEp.interopUseDtmfPayloadTypeFoundInAnswer="Value"

Machine Detection Configuration Software Configuration Guide

Figure 126: Payload Type Values
Value Meaning
0 disable
1 enable
Quantity of initial packets sent to transmit a DTMF Out-of-Band using RTP

You can specify the quantity of packets sent at the beginning of an Out-of-Band DTMF using RTP. This
variable also specifies the quantity of terminating packets that are sent at the end of the DTMF transmission.
Note that this variable has an effect only if the Transport Method drop-down menu is set to Out-of-band
using RTP (see “Miscellaneous Media Parameters” on page 263 for more details).
 To set the initial quantity of RTP packets:

1. In the miptMIB, set the InteropDtmfRtpInitialPacketQty variable with the proper quantity.
mipt.interopDtmfRtpInitialPacketQty="Value"
where Value may be between 1 and 3.
Machine Detection Configuration
The Machine Detection section allows you to set the tone detection parameters of the Aastra unit.
 To set Machine detection parameters:

2. In the Machine Detection section of the Misc page, select whether or not you want to override the
machine detection parameters set in the Default configuration in the Endpoint Specific drop-down
menu.
Figure 127: Machine Detection Section
2
3
4
5
6

Chapter 31 - Miscellaneous Media Parameters Machine Detection Configuration
3. Select whether or not you want to enable fax calling tone (CNG tone) detection in the CNG Tone
Detection drop-down menu.
Table 218: CNG Tone Detection Settings
Setting Description
Enable Upon recognition of the CNG tone, the unit switches the communication from
voice mode to fax mode and the CNG is transferred by using the preferred fax
codec.
Note: This option allows for quicker fax detection, but it also increases the risk
of false detection.
Disable The CNG tone does not trigger a transition from voice to data and the CNG is
transferred in the voice channel.
Note: With this option, faxes are detected later, but the risk of false detection
is reduced.
4. Select whether or not you want to enable CED tone detection in the CED Tone Detection drop-down
menu.
Table 219: CNG CED Detection Settings
Setting Description
Enable Upon recognition of the CED tone, the unit behaves as defined in the
Behavior on CED Tone Detection parameter Step 6).
Disable The CED tone does not trigger a transition to fax or voiceband data mode.
The CED is transferred in the voice channel.
5. Select whether or not you want to enable fax V.21 modulation detection in the V.21 Modulation
Table 220: V.21 Modulation Detection Settings
Setting Description
Enable Upon recognition of the V.21 modulation tone, the unit switches the
communication from voice mode to fax mode and the signal is transferred by
using the preferred fax codec.
Disable The V.21 modulation does not trigger a transition from voice to data and the
signal is transferred in the voice channel.
6. Define the behaviour of the unit upon detection of a CED tone in the Behavior on CED Tone
Table 221: CED Tone Detection Settings
Setting Description
Passthrough The CED tone triggers a transition from voice to voice band data and is
transferred in the voice channel.. Use this setting when any kind of analog
device (i.e.: telephone, fax or modem) can be connected to this port.
Fax Mode Upon detection of a CED tone, the unit switches the communication from
voice mode to fax mode and the CED is transferred by using the preferred fax
codec. Only a fax can then be connected to this port.
Note: This parameter has no effect if the CED Tone Detection parameter is set to Disabled.

Base Ports Configuration Software Configuration Guide
Base Ports Configuration
The Base Ports section allows you to set the ports that the Aastra unit uses for different transports.
This section is available only in the default endpoint configuration.
 To set base ports parameters:

2. In the Base Ports section of the Misc page, set the UDP port number you want to use as RTP/RTCP
base port in the RTP field.
The RTP/RTCP ports are allocated starting from this base port.
RTP ports number are even and RTCP ports number are odd.
The default RTP/RTCP base port is 5004. For instance, assuming that the base port is defined on
5004, if there is currently no ongoing call and there is an incoming or outgoing call, the unit uses the
RTP/RTCP ports 5004 and 5005.
Figure 128: Base Ports Section
2
3
4
3. Set the UDP port number you want to use as SRTP/SRTCP base port in the SRTP field.
The SRTP/SRTCP ports are allocated starting from this base port.
SRTP ports number are even and SRTCP ports number are odd.
The default SRTP/SRTCP base port is 5004. For instance, assuming that the base port is defined
on 5004, if there is currently no ongoing call and there is an incoming or outgoing call, the unit uses
the SRTP/SRTCP ports 5004 and 5005.
Using the same base port for RTP/RTCP and SRTP/SRTCP does not conflict.
Note that if the media transport is set to “Secure with fallback” (“Chapter 32 - Security” on page 377),
both RTP and SRTP base ports are used at the same time when initiating an outgoing call. If there
is currently no call and the default base ports are used, the RTP port is 5004 and the SRTP port is
the next available port starting from the base port, which is 5006.
4. Set the port number you want to use as T.38 base port in the T.38 field.
The T.38 ports are allocated starting from this base port.
The default T.38 base port is 6004. For instance, assuming that the base port is defined on 6004 if
there is currently no ongoing call and there is an incoming or outgoing call, the unit uses the T.38
port 6005.

Chapter 31 - Miscellaneous Media Parameters Base Ports Configuration

AastraSoftware
Configuration Guide
Telephony Parameters
C H A P T E R
32 DTMF Maps Configuration
This chapter describes how to configure and use the DTMF maps of the Aastra unit.
Standards Supported • RFC 2705: Media Gateway Control Protocol (MGCP) Version
1.0, section 3.4 (Formal syntax description of the protocol).
 DTMF maps syntax.

 General DTMF maps parameters.
 Allowed DTMF maps parameters.
 Refused DTMF maps parameters.
Introduction
A DTMF map (also called digit map or dial map) allows you to compare the number users just dialed to a string
of arguments. If they match, users can make the call. If not, users cannot make the call and get an error signal.
It is thus essential to define very precisely a DTMF map before actually implementing it, or your users may
encounter calling problems.
Because the Aastra unit cannot predict how many digits it needs to accumulate before transmission, you could
use the DTMF map, for instance, to determine exactly when there are enough digits entered from the user to
place a call.
Syntax
The permitted DTMF map syntax is taken from the core MGCP specification, RFC 2705, section 3.4:
DigitMap = DigitString / '(' DigitStringList ')'
DigitStringList = DigitString 0*( '|' DigitString )
DigitString = 1*(DigitStringElement)
DigitStringElement = DigitPosition ['.']
DigitPosition = DigitMapLetter / DigitMapRange
DigitMapLetter = DIGIT / '#' / '*' / 'A' / 'B' / 'C' / 'D' / 'T'
DigitMapRange = 'x' / '[' 1*DigitLetter ']'
DigitLetter ::= *((DIGIT '-' DIGIT ) / DigitMapLetter)
Where “x” means “any digit” and “.” means “any number of”.
For instance, using the telephone on your desk, you can dial the following numbers:
Table 222: Number Examples
Number Description
0 Local operator
00 Long distance operator
xxxx Local extension number
8xxxxxxx Local number
#xxxxxxx Shortcut to local number at other corporate sites
91xxxxxxxxxx Long distance numbers

Chapter 32 - DTMF Maps Configuration Syntax
Table 222: Number Examples (Continued)
Number Description
9011 + up to 15 digits International number
The solution to this problem is to load the Aastra unit with a DTMF map that corresponds to the dial plan.
A Aastra unit that detects digits or timers applies the current dial string to the DTMF map, attempting a match
to each regular expression in the DTMF map in lexical order.
 If the result is under-qualified (partially matches at least one entry in the DTMF map), waits for
more digits.
 If the result matches, dials the number.
 If the result is over-qualified (i.e., no further digits could possibly produce a match), sends a fast
busy signal.
Special Characters
DTMF maps use specific characters and digits in a particular syntax.
Table 223: DTMF Map Characters
Character Use
Digits (0, 1, 2... 9) Indicates specific digits in a telephone number expression.

T The Timer indicates that if users have not dialed a digit for the time defined, it is likely
that they have finished dialing and the SIP Server can make the call.
x Matches any digit, excluding “#” and “*”.
| Indicates a choice of matching expressions (OR).
. Matches an arbitrary number of occurrences of the preceding digit, including 0.
[ Indicates the start of a range of characters.
] Indicates the end of a range of characters.
How to Use a DTMF Map

Let’s say you are in an office and you want to call a co-worker’s 3-digits extension. You could build a DTMF
map that says “after the user has entered 3 digits, make the call”. The DTMF map could look as follows:
xxx
You could refine this DTMF map by including a range of digits. For instance, you know that all extensions in
your company either begin with 2, 3, or 4. The corresponding DTMF map could look as follows:
[2-4]xx
If the number you dial begins with anything other than 2, 3, or 4, the call is not placed and you get a busy signal.
Combining Several Expressions

You can combine two or more expressions in the same DTMF map by using the “|” operator, which is equal to
OR.
Let’s say you want to specify a choice: the DTMF map is to check if the number is internal (extension), or
external (a local call). Assuming that you must first dial “9” to make an external call, you could define a DTMF
map as follows:
([2-4]xx|9[2-9]xxxxxx)
The DTMF map checks if:
 the number begins with 2, 3, or 4 and
 the number has 3 digits
If not, it checks if:

Syntax Software Configuration Guide
 the number begins with 9 and

 the second digit is any digit between 2 and 9 and
 the number has 7 digits
Note: Enclose the DTMF map in parenthesis when using the “|” option.
Using the # and * Characters

It may sometimes be required that users dial the “#” or “*” to make calls. This can be easily incorporated in a
DTMF map:
xxxxxxx#
xxxxxxx*
The “#” or “*” character could indicate users must dial the “#” or “*” character at the end of their number to
indicate it is complete. You can specify to remove the “#” or “*” found at the end of a dialed number. See
“General DTMF Maps Parameters” on page 282.
Using the Timer

The Timer indicates that if users have not dialed a digit for the time defined, it is likely that they have finished
dialing and the Aastra unit can make the call. A DTMF map for this could be:
[2-9]xxxxxxT
Note: When making the actual call and dialing the number, the Aastra unit automatically removes the “T”
found at the end of a dialed number, if there is one (after a match). This character is for indication purposes
only.
See “General DTMF Maps Parameters” on page 282 for more details.
Calls Outside the Country

If your users are making calls outside their country, it may sometimes be hard to determine exactly the number
of digits they must enter. You could devise a DTMF map that takes this problem into account:
001x.T
In this example, the DTMF map looks for a number that begins with 001, and then any number of digits after
that (x.).
Example
Table 222 on page 279 outlined various call types one could make. All these possibilities could be covered in
one DTMF map:
(0T|00T|[1-7]xxx|8xxxxxxx|#xxxxxxx|91xxxxxxxxxx|9011x.T)
Validating a DTMF Map

The Aastra unit validates the DTMF map as you are entering it and it forbids any invalid value.

Chapter 32 - DTMF Maps Configuration General DTMF Maps Parameters
General DTMF Maps Parameters
The following are the general DTMF maps parameters you can set.
 To set the general DTMF map parameters:

1. In the web interface, click the Telephony link, then the DTMF Maps sub-link.
Figure 129: Telephony – DTMF Maps Web Page
2 3 4 5
2. In the General Configuration section, define the time, in milliseconds (ms), between the start of the
dial tone and the receiver off-hook tone, if no DTMF is detected, in the First DTMF Timeout field.
Values range from 1000 ms to 180000 ms. The default value is 20000 ms.
If you want to set a different First DTMF Timeout value for one or more endpoints, click the Edit
Endpoints button (see “Configuring Timeouts per Endpoint” on page 283 for more details).
3. Define the value, in milliseconds (ms), of the “T” digit in the Inter Digit Timeout field.
The “T” digit expresses a time lapse between the detection of two DTMFs. Values range from 500
ms to 10000 ms. The default value is 4000 ms.
If you want to set a different Inter Digit Timeout value for one or more endpoints, click the Edit
4. Define the total time, in milliseconds (ms), the user has to dial the DTMF sequence in the
Completion Timeout field.
The timer starts when the dial tone is played. When the timer expires, the receiver off-hook tone is
played. Values range from 1000 ms to 180000 ms. The default value is 60000 ms.
If you want to set a different Completion Timeout value for one or more endpoints, click the Edit
5. In the DTMF Maps Digit Detection (FXO/FXS) drop-down menu, define when a digit is processed
through the DTMF maps.
This parameters is available only when the unit has FXS or FXO ports.
Table 224: DTMF Maps Digit Detection Parameters
When Digits are processed as soon as they are pressed. This can lead to a digit leak in
Pressed the RTP at the beginning of a call if the voice stream is established before the last
digit is released.
When Digits are processed only when released. This option increases the delay needed
Released to match a dialed string to a DTMF map. There is also an impact on the First DTMF
Timeout, Inter Digit Timeout and Completion Timeout parameters since the timers
are stopped at the end of a digit instead of the beginning.

Allowed DTMF Maps Software Configuration Guide
Configuring Timeouts per Endpoint

You can set a different timeout value for one or more endpoints.
 To set a different value per endpoint:

1. In the General Configuration section of the DTMF Maps page, click the Edit Endpoints button.
The following window is displayed:
Figure 130: DTMF Map Timeout Section
2. Set the Override drop-down menu for the endpoint you want to set to Enable.
3. Change the value of one or more timeouts as required.
4. Repeat for each endpoint that you want to modify.
5. Click Submit when finished.
Allowed DTMF Maps
You can create/edit ten DTMF maps for the Aastra unit. DTMF map rules are checked sequentially. If a
telephone number potentially matches two of the rules, the first rule encountered is applied.
 To set up DTMF maps:

1. In the DTMF Map drop-down menu at the top of the window, select Allowed.
The Allowed DTMF Map section displays.
2. In the Allowed DTMF Map section – Enable column, enable one or more DTMF maps by selecting
the corresponding Enable choice.
Figure 131: Allowed DTMF Map Section

2 3 4 5 6 7 8

Chapter 32 - DTMF Maps Configuration Allowed DTMF Maps
3. Select the entity to which apply the allowed DTMF map in the Apply to column.
Table 225: DTMF Map Entity
Unit The DTMF map entry applies to the unit.

Endpoint The DTMF map applies to a specific endpoint. The endpoint is specified in the
Endpoint column of the same row.
specific row.
You can specify more than one endpoint. In that case, the endpoints are separated with a comma
(,). You can use the Suggestions column’s drop-down menu to select between suggested values, if
any.
5. Define the DTMF map string that is considered valid when dialed in the DTMF Map column.
The string must use the syntax described in “DTMF Maps Configuration” on page 279. A DTMF map
string may have a maximum of 64 characters.
6. Enter the DTMF transformation to apply to the signalled DTMFs before using it as call destination
in the Transformation column.
The following are the rules you must follow; “x” represents the signalled number.
• Add before “x” the DTMF to prefix or/and after “x” the suffix to add. Characters
“0123456789*# ABCD” are allowed.
• Use a sequence of DTMFs between “{}” to remove a prefix/suffix from the dialed
number if present. Use before “x” to remove a prefix and after “x” to remove a suffix.
Characters “0123456789*#ABCD” are allowed.
• Use a number between “()” to remove a number of DTMFs. Use before “x” to remove
DTMFs at the beginning of the number and after “x” to remove DTMFs at the end.
Characters “0123456789” are allowed.
The transformations are applied in order from left to right.
The following table gives an example with “18195551111#” as signalled number.
Table 226: DTMF Map Transformation Examples
Action Transformation Result
Add the prefix “0” to the dialed number 0x 018195551111#

Remove the suffix “#” from the dialed x{#} 18195551111
number
Remove the first four DTMFs from the dialed (4)x 5551111#
number
Remove the international code and (1){819}514x{#} 5145551111
termination and replace the area code by
another one
Replace the signalled DTMFs by “3332222” 3332222 3332222
7. Define the target to use when the DTMF map matches in the Target column.
This allows associating a target (FQDN) with a DTMF map. This defines a destination address to
use when the DTMF map matches. This address is used as destination for the INVITEs in place of
the “home domain proxy”. This is useful for such features as the speed dial and emergency call.
The default target is used when the value is empty.
The dialed DTMFs are not used if the target contains a user name.

Refused DTMF Maps Software Configuration Guide
8. Enable/Disable the emergency process of the call in the Emergency column.

• Disable: The call is processed normally.
• Enable: The call is processed as emergency.
The Emergency Call service (also called urgent gateway) allows a “911”-style service. It allows a
user to dial a special DTMF map resulting in a message being sent to a specified urgent gateway,
bypassing any other intermediaries.
If enabled, whenever the user dials the specified DTMF map, a message is sent to the target
address.
Refused DTMF Maps
A refused DTMF map forbids to call specific numbers; for instance, you want to accept all 1-8xx numbers
except 1-801. You can create/edit ten refused DTMF maps for the Aastra unit.
A refused DTMF map applies before an allowed DTMF map.
 To set up refused DTMF maps:

1. In the DTMF Map drop-down menu at the top of the window, select Refused.
The Refused DTMF Map section displays.
2. In the Refused DTMF Map section – Enable column, enable one or more DTMF maps by selecting
the corresponding Enable choice.
Figure 132: Refused DTMF Map Section

2 3 4 5
3. Select the entity to which apply the refused DTMF map in the Apply to column.
Table 227: DTMF Map Entity
Unit The DTMF map entry applies to the unit.

Endpoint The DTMF map applies to a specific endpoint. The endpoint is specified in the
Endpoint column of the same row.
specific row.

Chapter 32 - DTMF Maps Configuration Refused DTMF Maps
You can specify more than one endpoint. In that case, the endpoints are separated with a comma
(,). You can use the Suggestions column’s drop-down menu to select between suggested values, if
any.
5. Define the DTMF map string that is considered valid when dialed in the DTMF Map column.
The string must use the syntax described in “DTMF Maps Configuration” on page 279. A DTMF map
string may have a maximum of 64 characters.

C H A P T E R
33 Call Forward Configuration
This chapter describes how to set three types of Call Forward:

 On Busy
 On No Answer
 Unconditional
Note: This web page is available only on the following models: TA7102i
Call Forward On Busy
You can automatically forward the incoming calls of your users to a pre-determined target if they are already
on the line. The user does not have any feedback that a call was forwarded.
You can enable the Call Forward On Busy feature in two ways:
 By allowing the user to configure the call forward activation and its destination via the handset
(Steps 4-6).
 By manually enabling the service (Steps 7-8).
 To set the Call Forward On Busy feature:

1. In the web interface, click the Telephony link, then the Call Forward sub-link.
Figure 133: Telephony – Call Forward Web Page
3
4
5
6
7
8
2. Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at
the top of the window.
You have the choice between Default and all FXS endpoints your Aastra unit has.
3. In the Call Forward On Busy section, define whether or not you want to override the Call Forward
On Busy parameters set in the Default configuration in the Endpoint Specific drop-down menu.

Chapter 33 - Call Forward Configuration Call Forward On Busy
4. Enable the Call forward configuration via handset service by setting the Allow Activation via
Handset drop-down menu to Enable.
You also need to configure the activation and deactivation DTMF maps (steps 5 and 6).
If you select Disable, this does not disable the call forward, but prevents the user from activating or
deactivating the call forward service. The user will not be able to use the digits used to activate and
deactivate the call forward service.
5. Define the digits that users must dial to start the service in the DTMF Map Activation field.
This field is available only in the Default configuration.
For instance, you could decide to put “*72” as the sequence to activate the service. This sequence
must be unique and follow the syntax for DTMF maps (see “Chapter 35 - DTMF Maps Configuration”
on page 401). Dialing this DTMF map does not have any effect unless the service’s status is
“enabled”.
The activating sequence is set for all the endpoints of the Aastra unit. You cannot have a different
sequence for each endpoint.
6. Define the digits that users must dial to stop the service in the DTMF Map Deactivation field.
For instance, you could decide to put “*73” as the sequence to deactivate the service. This
sequence must be unique and follow the syntax for DTMF maps (see “Chapter 35 - DTMF Maps
Configuration” on page 401). Dialing this DTMF map does not have any effect unless the service’s
status is “enabled”.
The deactivating sequence is set for all the endpoints of the Aastra unit. You cannot have a different
7. Set the call forward service in the Activation field to Inactive or Active.
Table 228: Activation State
State Description
Inactive The call forward service is not available on the telephone connected to the specific
endpoint. A call to this endpoint is not forwarded if the endpoint is busy.
Active The call forward service is available on the telephone connected to the specific
endpoint. A call to the endpoint is forwarded to the specified destination if the
endpoint is busy. You must define the call forward destination in the Forwarding
Address field (Step 8). The call forward service behaves as if it is inactive if the
Forwarding Address is empty.
To let the user activate or deactivate this service with his or her handset, see steps 4, 5, and 6. In
that case, the field is automatically updated to reflect the activation status.
8. Define the address to which forward incoming calls in the Forwarding Address field.
Accepted formats are:
• telephone numbers (5551111)
• SIP URLs such as ”scheme:user@host”. For instance, “sip:[email protected]”.
This string is used literally, so cosmetic symbols (such as the dash in “555-xxxx”) should not be
present.
Configuring Call Forward on Busy via Handset

The following is the procedure to use this service on the user’s telephone.

Call Forward On No Answer Software Configuration Guide
 To forward calls:
1. Take the receiver off-hook.
2. Wait for the dial tone.
3. Dial the sequence implemented to activate the call forward on busy service.
This sequence could be something like *72.
4. Wait for the stutter dial tone (three “beeps”) followed by the dial tone.
5. Dial the number to which you want to forward your calls. Dial any access code if required.
6. Wait for three “beeps” followed by a silent pause.
The call forward is established.
7. Hang up your telephone.
 To cancel the call forward:

3. Dial the sequence implemented to deactivate the call forward on busy service.
4. Wait for the transfer tone (three “beeps”) followed by the dial tone.
The call forward is cancelled.
Call Forward On No Answer
You can forward the incoming calls of your users to a pre-determined target if they do not answer their
telephone before a specific amount of time. The user does not have any feedback that a call was forwarded.
(Steps 3-5).
 To set the Call Forward On No Answer feature:

2. In the Call Forward On No Answer section, define whether or not you want to override the Call
Forward On No Answer parameters set in the Default configuration in the Endpoint Specific drop-
down menu.
Figure 134: Telephony – Call Forward on No Answer section
2
3
4
5
6
7
8

Chapter 33 - Call Forward Configuration Call Forward On No Answer
“enabled”.
6. Define the time, in milliseconds, the telephone keeps ringing before the call forwarding activates in
the Timeout field.
7. Set the status of the service in the Activation field to Inactive or Active.
State Description
present.
Configuring Call Forward on Answer via Handset


Call Forward Unconditional Software Configuration Guide
3. Dial the sequence implemented to activate the call forward on no answer service.

3. Dial the sequence implemented to deactivate the call forward on no answer service.
Call Forward Unconditional
The Call Forward Unconditional feature allows users to forward all of their calls to another extension or line.
(Steps 3-5).
 To set the Call Forward Unconditional feature:

2. In the Unconditional section, define if you want to override the Call Forward Unconditional
parameters set in the Default configuration in the Endpoint Specific drop-down menu.
Figure 135: Telephony – Call Forward Unconditional Section
2
3
4
5
6
7

Chapter 33 - Call Forward Configuration Call Forward Unconditional
“enabled”.
6. Set the status of the service in the Activation field to Inactive or Active.
State Description
present.
Configuring Call Forward on Unconditional via Handset

When forwarding calls outside the system, a brief ring is heard on the telephone to remind the user that the
call forward service is active. The user can still make calls from the telephone.

Call Forward Unconditional Software Configuration Guide
3. Dial the sequence implemented to activate the call forward unconditional service.
 To check if the call forward has been properly established:

3. Dial your extension or telephone number.
The call is forwarded to the desired telephone number.

3. Dial the sequence implemented to deactivate the call forward – unconditional service.

Chapter 33 - Call Forward Configuration Call Forward Unconditional

C H A P T E R
34 Telephony Services
Configuration
This chapter describes how to set the following subscriber services:

 Hook Flash Processing
 Automatic call
 Call completion
 Delayed Hotline
 Call Transfer
 Call Waiting
 Conference
 Direct IP address call
 Hold
 Second call
 Message Waiting Indicator
Some of the subscriber services are not supported on all Aastra unit models, so your specific model may not
have all subscriber services listed in this chapter.
Standards Supported • RFC 2976: The SIP INFO Method
The General Configuration sub-section of the Services Configuration section allows you to define the Hook
Flash Processing feature.
Note: Performing a flash hook and pressing the flash button means the same thing. However, not all
telephone models have a flash button.

Chapter 34 - Telephony Services Configuration General Configuration
 To set general services parameters:

1. In the web interface, click the Telephony link, then the Services sub-link.
Figure 136: Telephony – Services Web Page
3
4
3. In the General Configuration sub-section, define whether or not you want to override the general
services parameters set in the Default configuration in the Endpoint Specific drop-down menu.
4. Select how to process hook-flash detection in the Hook Flash Processing drop-down menu.
Hook flash processing allows hook flash signals to be transported over the IP network allowing to
use advanced telephony services. Users normally press the “flash” button of the telephone during
a call in progress to put this call on hold, transfer it, or even initiate a conference call.
You can define whether these subscriber services are handled by the unit or delegated to a remote
party. If services are to be handled by a remote party, a SIP INFO message is sent to transmit the
user's intention.
Note: The hook-flash processing attribute is not negotiated in SDP.
Table 231: Hook Flash Settings
Setting Definition
Process Locally The hook-flash is processed locally. The actual behaviour of the “flash”
button depends on which endpoint services are enabled for this
endpoint.
Transmit Using The hook-flash is processed by a remote party. The hook-flash event is
Signaling Protocol carried by a signaling protocol message. The actual behaviour of the
“flash” button depends on the remote party.
The hook-flash event is relayed as a SIP INFO message as described
in RFC 2976.

General Configuration Software Configuration Guide
Automatic Call
The automatic call feature allows you to define a telephone number that is automatically dialed when taking
the handset off hook.
When this service is enabled, the second line service is disabled but the call waiting feature is still functional.
The user can still accept incoming calls.
 To set the automatic call feature:

2. In the Automatic Call sub-section, define whether or not you want to override the automatic call
Figure 137: Telephony – Automatic Call Section
2
3
4
3. Enable the service by setting the Automatic Call Activation drop-down menu to Enable.
4. Define the string to dial when the handset is taken off hook in the Automatic Call Target field.
present.
Call Completion
Standards Supported • RFC 4235: An INVITE-Initiated Dialog Event Package for the
Session Initiation Protocol (SIP)a
• draft draft-poetzl-bliss-call-completion-00b
a. Implemented in client mode only and used for the call completion.
b. Implement the solution 1 in 5.1.
The call completion service allows you to configure the Completion of Calls on No Reply (CCNR) and
Completion of Calls to Busy Subscriber (CCBS) features.
CCBS allows a caller to establish a call with a “busy” callee as soon as this callee is available to take the call.
It is implemented by monitoring the activity of a UA and look for the busy-to-idle state transition pattern.
CCNR allows a caller to establish a call with an “idle” callee right after this callee uses his phone. It is
implemented by monitoring the activity of a UA and look for the idle-busy-idle state transition pattern.
The information about the call completion is not kept after a restart of the EpServ service. This includes the
call completion activation in the Pots service and the call completion monitoring in the SipEp service.
 To set the call completion feature:


2. In the Call Completion sub-section, define whether or not you want to override the call completion
Figure 138: Telephony – Call Completion Section
3
4
5
6
7
8
9
10
11
12
13
3. Enable or disable the (CCBS) service by selecting the proper value in the Allow CCBS Activation
Via Handset drop-down menu.
4. If the CCBS service is enabled, define the digits that users must dial to start the service in the CCBS
DTMF Map Activation field.
You can use the same code in the CCNR DTMF Map Activation field.
“enabled”.
5. Enable or disable the (CCNR) service by selecting the proper value in the Allow CCNR Activation
Via Handset drop-down menu.
6. If the CCNR service is enabled, define the digits that users must dial to start the service in the CCNR
DTMF Map Activation field.
You can use the same code in the CCBS DTMF Map Activation field.
“enabled”.
7. Define the digits that users must dial to stop the CCBS and CCNR services in the DTMF Map
Deactivation field.
For instance, you could decide to put “*94” as the sequence to deactivate the services. This
8. Define the delay, in minutes, after the call completion activation to automatically deactivate the call
completion if the call is not completed in the Expiration Timeout field.


9. Select the call completion method to detect that the call completion destination is ready to complete
the call in the Method drop-down menu.
Table 232: Call Completion Method Parameters
Method Desciption
Monitoring Only The call completion only uses the monitoring method to detect that the
destination is ready to complete the call.
Monitoring And Polling The call completion only uses the monitoring method to detect that the
destination is ready to complete the call. The polling mechanism is used
if the call completion destination cannot be monitored.

The monitoring method consists of using the protocol signalling to detect the destination state
without using the call. When the destination is ready to complete the call, the local user is notified
that the call is ready to be completed and the call to the destination is initiated when the user is ready
to initiate the call.
The polling method consists of using periodic calls to the call completion destination until the
destination responds with a ringing or connect. Upon receiving these responses, the local user is
notified that the call is ready to be completed.
The polling mechanism can only be used for call completion to busy subscriber (CCBS).
The retransmission of the polling mechanism is configurable with
DefaultCallCompletionPollingInterval.
10. Enable or disable the call completion auto reactivation in the Auto Reactivate drop-down menu.
When enabled, the call completion busy subscriber is automatically activated if the call initiated by
a call completion busy subscriber or call completion no response fails because of a busy
destination.
11. Define the minimal delay to wait, in seconds, before executing a call completion after its activation
in the Auto Reactivate Delay field.
This delay only applies to call completion activated via the call completion auto reactivation feature
(See Step 9).
Aastra recommends to set a delay when the method to monitor the target state is based on the
target calls instead of its ability to answer a call.
If the timeout is set to 0 and the target is off hook, the FXS endpoint always rings to notify that the
call completion is ready to be completed. However the call is always busy and thus reactivated
without the possibility for the user to cancel the call completion. The call completion will continue
until the ringing or call completion timeout or if the target became ready to receive call.
12. Define how the call completion service needs to interpret the reception of a progress message with
early media in the Early Media Behaviour drop-down menu.
Table 233: Call Completion Early Media Behaviour Parameters
None The progress message with early media is not considered as a busy or a ringing
response.
CCBS The progress message with early media is interpreted as a busy response and the
CCBS can be activated on the call.
CCNR The progress message with early media is interpreted as a ringing response and
the CCNR can be activated on the call.


13. Define the delay, in seconds, between the calls to the call completion target used for the polling
mechanism in the Polling Interval field.
This parameter is used only if the Default Call Completion Method drop-down menu is set to
Monitoring And Polling.
Special SIP Configuration

If you are using an Asterisk® IP PBX, it returns the error code 503 instead of 486 for a busy destination when
the call limit is reached. The following error mapping can be required:
1. Go to the page SIP > Misc.
2. Insert a new mapping (with the plus button) in the SIP To Cause Error Mapping section.
3. Set the SIP code to 503 “Service Unavailable” and the cause to 17 “User busy”.
4. Click Submit.
Using the Call Completion Services

The following are the various procedures to use these services on the user’s telephone.
 To start the CCBS (procedure 1)

The call has reached a busy destination and the busy tone is played.
1. Dial the sequence implemented to enable the CCBS.
The confirmation tone is played.
2. Hang up the telephone.
Alternatively, you can use procedure 2.
 To start the CCBS (procedure 2)

The call has reached a busy destination and the busy tone is played.
The dial tone is played
3. Dial the sequence implemented to enable the CCBS.
Alternatively, you can use procedure 1.
 To start the CCNR

The call has reached a destination but the call is still not yet established. A ring back or welcome message is
generally played at this moment.

3. Dial the sequence implemented to enable the CCNR.

 To stop the CCBS or CCNR

2. Dial the sequence implemented to disable the CCBS and CCNR.
 When the call completion target is ready to receive a call:

1. The telephone rings with the distinctive ringing “Bellcore-dr2” (0.8 On – 0.4 Off, 0.8 On – 4.0 Off).
The call is initiated to the call completion destination.
Call Transfer
The Call Transfer service offers two ways to transfer calls:
 Blind Transfer
 Attended Transfer
 To enable the Call Transfer services:

2. In the Call Transfer sub-section, define whether or not you want to override the call transfer
Figure 139: Telephony – Call Transfer Web Page
2
3
4
3. Enable the Blind Transfer service by setting the Blind Transfer Activation drop-down menu to
Enable.
The blind call transfer service is sometimes called Transfer without Consultation or Unattended
Transfer. It allows a user to transfer a call on hold to a still ringing (unanswered) call. The individual
at the other extension or telephone number does not need to answer to complete the transfer.
The call hold and second call services must be enabled for this service to work. See “Call Hold” on
page 311 and “Second Call” on page 311.
4. Enable the Attended Transfer service by setting the Attended Transfer Activation drop-down menu
to Enable.
The attended call transfer service is sometimes called Transfer with Consultation. It allows a user
to transfer a call on hold to an active call. The individual at the other extension or telephone number
must answer to complete the transfer.

The call hold and second call services must be enabled for this service to work. See “Call Hold” on
page 311 and “Second Call” on page 311.
Using Blind Call Transfer

To configure the SIP Blind Transfer Method, see “SIP Blind Transfer Method” on page 345.
 To transfer a current call blind:

1. Perform a Flash-Hook by pressing the “Flash” button on your analog telephone.
This puts the call on hold.
2. Wait for the transfer tone (three “beeps”).
3. Dial the number to which you want to transfer the call.
4. Wait for the ringback tone, then hang up your telephone.
The call is transferred.
Once the transfer is executed, the remaining calls (call on hold and ringing call with third party) are
then connected together. The call on hold is automatically unheld and hears the ringback tone
provided by the third party's ringing.
You can also wait for the third party to answer if you want. In this case, the call transfer becomes
attended.
If you want to get back to the first call (the call on hold), you must perform a Flash-Hook.
You are back with the first call and the third party is released.
Using Attended Call Transfer

 To transfer a current call attended:

This puts the call on hold.
2. Wait for the transfer tone (three “beeps”).
3. Dial the number to which you want to transfer the call.
The third party answers.
The call is transferred.
5. If you want to get back to the first call (the call on hold), you must perform a Flash-Hook before the
target answers.
You are back with the first call and the third party is released.
Note: If the number to which you want to transfer the call is busy or does not answer, perform a Flash-Hook.
The busy tone or ring tone is cancelled and you are back with the first call.
Call Waiting
The call waiting tone indicates to an already active call that a new call is waiting on the second line.
Your users can activate/deactivate the call waiting tone for their current call. This is especially useful when
transmitting faxes. The user that is about to send a fax can thus deactivate the call waiting tone to ensure that
the fax transmission will not be disrupted by an unwanted second call. When the fax transmission is completed
and the line is on-hook, the call waiting tone is automatically reactivated.

 To set the Call Waiting services:

2. In the Call Waiting sub-section, define whether or not you want to override the call waiting
Figure 140: Call Waiting Section
2
3
4
3. Enable the service by setting the Call Waiting Activation drop-down menu to Enable.
This permanently activates the call waiting tone. When receiving new calls during an already active
call, a special tone is heard to indicate that a call is waiting on the second line. The user can then
answer that call by using the “flash” button. The user can switch between the two active calls by
using the “flash” button.
The call hold service must be enabled for this service to work. See “Call Hold” on page 311.
If the user is exclusively using faxes, select Disable to permanently disable the call waiting tone.
4. Define the digits that users must dial to disable the Call Waiting tone in the Cancel DTMF Map field.
This allows a user who has call waiting enabled to disable that service on the next call only. If, for
any reason, the user wishes to undo the cancel, unhook and re-hook the telephone to reset the
service.
For instance, you could decide to put “*76” as the sequence to disable the call waiting tone. This
Using Call Waiting

The call waiting feature alerts the user if he or she is already on the telephone and a second call happens. A
“beep” (the call waiting tone) is heard and repeated every ten seconds to indicate there is a second incoming
call.
 To put the current call on hold:

This puts the call on hold and the second line is automatically connected to your line.
2. Answer the call on the second line.
 To switch from one line to the other:

1. Perform a Flash-Hook each time you want to switch between lines.
 To terminate the first call before answering the second call:

2. Wait for the telephone to ring.
3. Answer the telephone.

The second call is on the line.
Removing the Call Waiting Tone

You can temporarily deactivate the call waiting tone indicating a call is waiting. This is especially useful when
transmitting faxes. If you are about to send a fax, you can thus deactivate the call waiting tone to ensure that
the fax transmission is not disrupted by an unwanted second call. When the fax transmission is completed and
the line is on-hook, the call waiting tone is automatically reactivated.
 To deactivate the call waiting tone:

3. Dial the sequence implemented to deactivate the call waiting tone.
The call waiting tone is disabled.
IMS-3GPP Communication Waiting

Upon receipt of a SIP INVITE with multipart/mixed content where a valid IMS communication waiting indicator
is correctly specified such as in this example:
INVITE sip:...
[...]
Content-Type: multipart/mixed;boundary=boundary1
[...]
--boundary1
Content-Type: application/vnd.3gpp.cw+xml
Content-Disposition: render;handling=optional
<?xml version="1.0"?>
<ims-cw xmlns="urn:3gpp:ns:cw:1.0">
<communication-waiting-indication/>
</ims-cw>
--boundary1
Content-Type: application/sdp
[...]
--boundary1--
The 180 Ringing response to this may contain a special header :
Alert-Info: <urn:alert:service:call-waiting>
that is appended if all of the following are true :
1. The INVITE contained the <communication-waiting-indication/> 3GPP option.
2. The destination endpoint supports call waiting.
3. The call waiting feature is enabled for this endpoint.
4. The endpoint is currently in an active state (not ringing, not on hold, not on hook).
There are no variables to control this behaviour, it is always activated.
This header could be used by the server to notify the 2nd caller that the destination is currently busy
in a call but was notified of this new incoming call.

Conference
Note: It is recommended to use the conferencing functionality provided in the MX-ONE.
The Conference Call service allows a user to link two or more calls together to form a single conversation,
called a conference.
 Only 3-way conferences are currently supported.
 A participant of the conference can put the conference on hold and attempt other calls. This
participant may then rejoin the conference at a later time by unholding it. The participant who
initiated the conference cannot put it on hold.
You must enable the call hold, second call and attended call transfer services for this service to work. See
“Call Hold” on page 311, “Second Call” on page 311, and “The Call Transfer service offers two ways to transfer
calls:” on page 301.
The following is a conference call flow example:
Figure 141: Conference Call Flow
User INVITE (G.729) User User

Agent Trying/ Ringing/200 OK Agent Agent
#2 ACK #1 #3
(B) (A) (C)
Flash Hook
INVITE (HOLD)
Trying /200 OK
ACK
INVITE (G.729 )
Trying/Ringing/200 OK
ACK
Flash Hook
Trying200 OK
ACK
INVITE (UNHOLD-G.729 )
Trying/200 OK
ACK
3-way Conference Call Established
DSP Limitation
The Aastra Ta7102i model suffer from a limitation of their DSPs. When using a codec other than G.711,
enabling Secure RTP (SRTP) and/or using conferences has an impact on the Aastra unit’s overall
performance as SRTP and conferences require CPU power. That is the reason why there is a limitation on the
lines that can be used simultaneously, depending on the codecs enabled and SRTP. This could mean that a
user picking up a telephone on these models may not have a dial tone due to lack of resources in order to not
affect the quality of ongoing calls. See “Security” on page 201 for more details on SRTP limitations.
The DSPs offer channels as resources to the Aastra unit. The Aastra unit is limited to two conferences per
DSP.
Please note that:
 One FXS line requires one channel.
 Each conference requires one additional channel
 The TA7102i has one DSP

A total of eight channels per DSP are available when using unsecure communication, to be used between the
FXS lines and up to two conferences.
A total of six channels per DSP are available when using SRTP, to be used between the FXS lines and up to
two conferences.
Enabling the Conference Call Feature

You must enable this service before your users can use it.
 To enable the Conference service:

2. In the Conference sub-section, define whether or not you want to override the conference
Figure 142: Conference Section
2
3
3. Enable the service by setting the Conference Activation drop-down menu to Enable.
Using an External Server for the Conference
Standards Supported • RFC 4579: Session Initiation Protocol (SIP) - Call Control -
Conferencing for User Agentsa
a. Partially compliant. Only call flows of sections 5.4 and 5.6 are supported. RFC 4575 is not supported.
The Aastra unit can use an external server to mix the media of the conference. This conference type requires
the configuration of an external server. Using this type of conference does not affect the number of
simultaneous calls supported. You can use this feature only if the Conference service is enabled (see
“Enabling the Conference Call Feature” on page 305 for more details).
 To use a server-based conference:

1. In the EpServMIB, specify how to manage the conference by setting the defaultConferenceType
variable to the proper value.
This configuration only applies to a conference initiated by one of the unit's endpoint.
EpServ.defaultConferenceType="Value"


Table 234: Conference Type Parameters
Value Parameter Description
100 Local The media of the conference is locally mixed by the unit. This
conference type does not require any special support of the call peer
or server. Using this type of conference can reduce the number of
simultaneous calls supported.
200 ConferenceSer The unit uses an external server to mix the media of the conference.
ver This conference type requires the configuration of an external server
(See Step 3). Using this type of conference does not affect the number
of simultaneous calls supported.
In Local mode, the number of participants is limited to the unit's model capacity. In
ConferenceServer mode, the number of participants is limited by the server's capacity.
2. If you want to set a different conference type for one or more endpoints, set the following variables:
• epSpecificConferenceEnableConfig variable for the specific endpoint you want to
• epSpecificConferenceType variable for the specific endpoint you want to configure to
the proper value.
EpServ.epSpecificConference.EnableConfig[Id="Specific_Endpoint"]="1"
EpServ.epSpecificConference.Type[Id="Specific_Endpoint"]="Type"
where:
• Specific_Endpoint is the number of the endpoint you want to configure.
• Value is the type as defined in Step 1.
3. If you have set the Conference type to ConferenceServer, in the SipEpMIB, set the
defaultConferenceType variable with the URI used in the request-URI of the INVITE sent to the
conference server as defined in RFC 4579.
SipEp.DefaultStaticConferenceServerUri="URI"
4. If you want to set a different URI for one or more endpoints, set the following variables:
• GwSpecificConferenceEnableConfig variable for the specific endpoint you want to
• GwSpecificConferenceServerUri variable for the specific endpoint you want to
EpServ.GwSpecificConference.EnableConfig[Id="Specific_Endpoint"]="1"
EpServ.GwSpecificConference.ServerUri[Id="Specific_Endpoint"]="URIValue"
where:
• URIValue is the URI you want to use.
Managing a Conference Call

If you are on the telephone with one person and want to conference with a third one, you can do so. In the
following examples, let’s assume that:
 “A” is the conference initiator.

 “B” is the person called on the first line.

 “C” is the person called on the second line.
 “D” is a fourth person that “A” wants to add to the conference in conferenceServer conference
type.
 “E” is a fifth person that “C” wants to add to the conference in conferenceServer conference
type.
 To initiate a three-way conference (“A” and “B” already connected):

1. “A” performs a Flash-Hook.
This puts “B” on hold and the second line is automatically connected. “A” hears a dial tone.
2. “A” dials “C’s” number.
“A” and “C” are now connected.
3. “A” performs another Flash-Hook.
The call on hold (“B”) is reactivated. “A” is now conferencing with “B” and “C”.
 “B” (or “C”) hangs up during the conference:

1. “B” (or “C”) hangs up during the conference.
The conference is terminated, but the call between “A” and “C” (or “B”) is not affected and they are
still connected.
 “A” (conference initiator) hangs up during the conference:

1. “A” hangs up.
The conference is terminated, both call "C" and "B" are also terminated.
 “A” wants to add a fourth member to the conference:

This is available only in the conferenceServer conference type.
1. “A” performs a Flash-Hook.
“A” hears a dial tone. The second line is automatically connected. “B” and “C” are still in conference.
2. “A” dials “D’s” number.
“A” and “D” are now connected.
3. “A” performs another Flash-Hook.
“A” is now conferencing with “B”, “C”, and “D”.
 "C" wants to add a fifth member to the conference:

This is available only in the conferenceServer conference type.
1. "C" performs a Flash-Hook.
"C" hears a dial tone. The second line is automatically connected. "A ", "B " and "D " are still in
conference.
2. "C" dials "E's" number.
"C" and "E" are now connected.
3. "C" performs another Flash-Hook.
"E" is now conferencing with "A ", "B", "C", and "D".
Delayed Hot Line

The delayed hot line feature (also called warm line) is used to make an automatic call to a specified address
on the two following conditions:
 When the user picks up the phone but does not dial any digit. The configured destination is

automatically called upon picking up the phone and after waiting for the configurable number
of seconds without dialling.
 When the user starts dialing but does not complete a valid number before the timeout set in the
Delayed Hotline Condition drop-down menu expires.
The condition on which the delayed hotline is activated is configurable. This feature thus places an automatic
call whenever the Delayed Hotline Condition timeout expires. It could be used as an alternative to the
emergency number (for instance, the 911 number in North America).
 To configure the basic delayed hot line feature:

2. In the Delayed Hotline sub-section, define whether or not you want to override the delayed hotline
Figure 143: Delayed Hotline Section
2
3
3. Enable the service by setting the Delayed Hotline Activation drop-down menu to Enable.
When the feature is disabled, a user picking up the phone but not pressing any telephone keys
hears the Receiver Off-Hook tone after the amount of time specified in the
digitMapTimeoutFirstDigit variable.
 To configure the delayed hotline activation condition:

1. In the Delayed Hotline sub-section, select the condition(s) that activate the delayed hotline in the
Delayed Hotline Condition drop-down menu.
Table 235: Delayed Hotline Conditions
FirstDtmfTimeout The delayed hotline is activated when the timeout configured

in the First DTMF Timeout field of the Telephony > DTMF
Maps page elapses (“General DTMF Maps Parameters” on
page 404).
InterDtmfOrCompletionTimeout The delayed hotline is activated when the timeout configured
in the Completion Timeout field of the Telephony > DTMF
Maps page elapses or when the DTMFs collection fails
because the Inter DTMF Timeout parameter elapses
(“General DTMF Maps Parameters” on page 404).

Table 235: Delayed Hotline Conditions (Continued)
AnyTimeout The delayed hotline is activated when the timeout configured

in the Completion Timeout field of the Telephony > DTMF
Maps page elapses and when the DTMFs collection fails
because the Inter DTMF Timeout parameter elapses
(“General DTMF Maps Parameters” on page 404).
 To configure the delayed hotline target:

1. In the Delayed Hotline sub-section, set the destination (address or telephone number) that is
automatically called in the Delayed Hotline field.

present.
Direct IP Address Call

The IP address call service allows a user to dial an IP address without the help of a SIP server. Using this
method bypasses any server configuration of your unit.
The user can dial an IP address and enter an optional telephone number. Note that the optional telephone
number is matched by using the same digit maps as a normal call.
The IP address call method can be used when a SCN user wants to reach a LAN endpoint.
 To set the direct IP call feature:

This menu is available only in the default endpoints configuration.
2. Enable the service by setting the Direct IP Address Call drop-down menu to Enable.
Figure 146: Telephony – Direct IP Address Call Section
Dialing an IP Address
 To make an IP address call:

1. Dial “**” (IP address prefix).
2. Dial the numerical digits of the IP address and use the “*” for the “.” of the IP address.
3. Dial “*” to terminate the IP address if you do not need to specify a phone number.

For instance, let’s say you want to reach a one-line access device or another LAN endpoint such as
an IP Phone with the IP address 192.168.0.23. You must then dial the following digits:
**192*168*0*23*
4. If you need to specify the phone number of a specific line, dial “#” to terminate the IP address.
5. Dial the telephone number of the specific line you want to reach.
For example, let’s say you want to reach the telephone connected to Line 2 of the Aastra unit with
the IP address 192.168.0.23. The phone number assigned to Line 2 of this Aastra unit is 1234. You
must then dial the following digits:
**192*168*0*23#1234
In this case, the Aastra unit sends an INVITE [email protected].
Call Hold
The Call Hold service allows the user to temporarily put an existing call on hold, usually by using the “flash”
button of the telephone. The user can resume the call in the same way.
You must enable this service for the following services to work properly:
 Call Waiting
 Second Call
 Blind Transfer
 Conference
 To enable the Call Hold service:

2. In the Hold sub-section, define whether or not you want to override the call hold parameters set in
the Default configuration in the Endpoint Specific drop-down menu.
Figure 147: Hold Section
2
3
3. Enable the service by setting the Hold Activation drop-down menu to Enable.
Using Call Hold

 To put the current call on hold:

This puts the call on hold. You can resume the call in the same way.
Second Call
The Second Call service allows a user with an active call to put the call on hold, and then initiate a new call on
a second line. This service is most useful with the transfer and conference services.
The call hold service must be enabled for this service to work. See “Call Hold” on page 311.
You must enable this service for the following services to work properly:
 Blind Transfer

Chapter 34 - Telephony Services Configuration Message Waiting Indicator
 Conference
 To enable the Second Call service:

2. In the Second Call sub-section, define whether or not you want to override the second call
Figure 148: Second Call Section
2
3
3. Enable the service by setting the Second Call Activation drop-down menu to Enable.
Using Second Call

 To use the second call service:

This puts the call on hold and the second line is automatically connected to your line.
2. Initiate the second call.
Message Waiting Indicator
Standards Supported • RFC 3265: Session Initiation Protocol (SIP)-Specific Event

Notificationa
• RFC 3842: The Message Summary and Message Waiting
Indication Event Package for the Session Initiation Protocol
(SIP)b
a. Supports receiving blind NOTIFY without subscribing. Sending blind NOTIFY is not supported.
b. Supports receiving blind NOTIFY without subscribing. Sending blind NOTIFY is not supported.
The Message Waiting Indicator (MWI) service alerts the user when new messages have been recorded on a
voice mailbox. It is enabled by default.
After the message is recorded, the server sends a message (SIP NOTIFY request) to the Aastra unit listing
how many new and old messages are available. The Aastra unit alerts the user of the new message in two
different ways:
 The telephone’s LED blinks (if present). A FSK signal is sent on the FXS line.
 A message waiting stutter dial tone replaces the normal dial tone when the user picks up the

Message Waiting Indicator Software Configuration Guide
FXS line.
Note: The message waiting state does not affect the Second Call feature. When in an active call,
performing a flash-hook to get access to the second line plays the usual dial tone.
The Aastra unit supports to receive SIP MWI notifications via SIP NOTIFY requests as defined in RFC 3842
but with the following limitations/diversions:
 In addition to the SIP event string "message-summary" (RFC 3842), the string "simple-
message-summary" is accepted. The significations of those strings are identical.
 In addition to the SIP content type string "simple-message-summary" (RFC 3842), the string
"message-summary" is accepted. The significations of those strings are identical.
 Support of message-summary is not advertised in the SIP REGISTER.
Note that received SIP NOTIFY with an event different than "message-summary" or "simple-message-
summary" is not interpreted as a valid MWI notification.
 To disable the Message Waiting Indicator service:

1. In the potsMIB, set the fxsDefaultMessageWaitingIndicatorActivation variable to the proper
value.
pots.fxsDefaultMessageWaitingIndicatorActivation="100"
If you want to reactivate the feature, use the following:

pots.fxsDefaultMessageWaitingIndicatoreActivation="Value"
Table 236: Message Waiting Indicator Parameters
100 Disabled The user is not alerted of messages awaiting attention.

200 Tone When messages are awaiting attention, the user is alerted by a
message waiting tone when picking up the handset.
300 Visual When messages are awaiting attention, the user is alerted by a Visual
Message Waiting Indicator such as a blinking LED on the phone.
400 ToneAndVisual When messages are awaiting attention, the user is alerted by a Visual
Message Waiting Indicator such as a blinking LED on the phone, and
a message waiting tone when picking up the handset.
2. If you want to set a different activation for one or more endpoints, set the following variables:
• fxsSpecificMessageWaitingIndicatorEnableConfig variable for the specific
endpoint you want to configure to enable.
• fxsSpecificMessageWaitingIndicatorActivation variable for the specific endpoint
you want to configure to the proper value.
pots.fxsSpecificMessageWaitingIndicator.EnableConfig[Id="Specific_Endpoint"]="1"

Chapter 34 - Telephony Services Configuration Emergency Call Override
pots.fxsSpecificMessageWaitingIndicator.Activation[Id="Specific_Endpoint"]="Valu
e"
where:
• Value is the activation as defined in Step 1.
Visual Message Waiting Indicator Type

You can configure how the Visual Message Waiting Indicator is sent on FXS lines.
 To configure the visual message waiting indicator type:

1. In the potsMIB, set the fxsDefaultVisualMessageWaitingIndicatorType variable to the proper
value.
pots.fxsDefaultVisualMessageWaitingIndicatorType="Value"

Table 237: Visual Message Waiting Indicator Type Parameters
100 Fsk A FSK signal is sent to activate the VMWI on the phone.
200 FskAndVoltage Both FSK signal and high voltage signal are used to activate the VMWI
on the phone.
Note: This parameter applies only to the following models: TA7102i
Emergency Call Override
You can override the set of services that are activated during an emergency call.
 To set the emergency call override:

1. In the epServMIB, set the defaultEmergencyCallOverride variable to the proper value.
epServ.defaultEmergencyCallOverride="Value"

Table 238: Emergency Call Override Parameters
100 NoOverride The set of services for emergency calls remains the same as
configured.
200 NoServices Ignores any service requiring a flash-hook. Call waiting and all other
related services are deactivated.

Call Statistics Software Configuration Guide
Call Statistics
This section describes how to access data available only in the MIB parameters of the Aastra unit. You can
display these parameters as follows:
The following are the call statistics the Aastra unit keeps. Statistics are updated at the end of each call.
Table 239: Call Statistics
IncomingCallsReceived Number of incoming IP calls received on the endpoint

since service start.
IncomingCallsAnswered Number of incoming IP calls answered on the endpoint
IncomingCallsConnected Number of incoming IP calls that successfully completed
call setup signaling on the endpoint since service start.
IncomingCallsFailed Number of incoming IP calls that failed to complete call
setup signaling on the endpoint since service start.
OutgoingCallsAttempted Number of outgoing IP calls attempted for the endpoint
OutgoingCallsAnswered Number of outgoing IP calls answered by the called
party for the endpoint since service start.
OutgoingCallsConnected Number of outgoing IP calls that successfully completed
call setup signaling for the endpoint since service start.
OutgoingCallsFailed Number of outgoing IP calls that failed to complete call
setup signaling for the endpoint since service start.
CallsDropped Number of IP calls, on the endpoint since service start,
that were successfully connected (incoming or
outgoing), but dropped unexpectedly while in progress
without explicit user termination.
TotalCallTime Cumulative duration of all IP calls on the endpoint since
service start, in seconds.
 To display call statistics:

1. In the epServMIB, go to the CallStatistics table.
get epServ.callStatistics
 To reset call statistics values to zero:

1. In the epServMIB, set callStatistics.Reset to Reset for the endpoint to reset.
set epServ.callStatistics.Reset=Reset
2. In the epServMIB, set callStatistics[EplId=callStatisticsEpId].Reset to Reset to reset

only one specific endpoint.
where:
• callStatisticsEpId is the string that identifies the combination of an endpoint and a
channel. The endpoint name is the same as the EpId used to refer to endpoints in other

Chapter 34 - Telephony Services Configuration Default Outbound Priority Call Routing
tables. On endpoints with multiple channels, the channel number must be appended at
the end of the endpoint name, separated with a dash.
set epServ.callStatistics[EplId=callStatisticsEpId].Reset=Reset
Examples:
Slot3/E1T1-12 refers to endpoint Slot3/E1T1, channel 12.
Phone-Fax1 refers to FXS endpoint Phone-Fax1 on a 4102s.
Port06 refers to FXS endpoint Port06 on 4108/4116/4124.
No channel number is appended to FXS endpoint strings because FXS lines do not support multiple
channels.
Default Outbound Priority Call Routing
You can define how to route priority calls including emergency calls.
 To set the default outbound priority call routing:

1. In the sipEpMIB, set the defaultOutboundPriorityCallRouting variable to the proper value.
sipEp.defaultOutboundPriorityCallRouting="Value"

Table 240: Default Outbound Priority Call Routing Parameters
100 Normal Sends the call using normal SIP call routing to the outbound proxy (if
defined) and to the target host (usually the SIP server).
200 SkipOutbound Sends the call directly to the configured server skipping the outbound
Proxy proxy.

C H A P T E R
35 Tone Customization Parameters

Configuration
This chapter describes how to override the pattern for a specific tone defined for the selected country (see
“Appendix A - Country-Specific Parameters” on page 603 for more details). It covers the following topics:
 Current Tone Definition
 Tone Override
Current Tone Definition
The Tone Customization page allows you to both see the current definition and override the pattern of the
following tones:
 Busy  Message Waiting
 Call Waiting  Preemption
 Confirmation  Reorder
 Congestion  Ringback
 Dial  Receiver Off Hook (ROH)
 Hold  Special Information Tone (SIT)
 Intercept  Stutter
This includes the number of frequencies used, the tone value in Hertz (Hz), its power in dBm, as well as the
states configured.
 To see the current definition of a tone:

1. In the web interface, click the Telephony link, then the Tone Customization sub-link.
Figure 149: Telephony – Tone Customization Web Page
2. Select the proper tone to see in the Select Tone drop-down menu at the top of the window.

Chapter 35 - Tone Customization Parameters Configuration Tone Override
The Current Tone Definition and Current Tone States sections describe the current definition of the
selected tone.
Tone Override
You can override the pattern for a specific tone. This is done in two sections:
Table 241: Tone Override Sections
State Description
Overridden Tone Definition Allows you to define up to four frequencies (F1 to F4). You must enter at least
one frequency.
Overridden Tone States Description of the tone state. You can define up to eight states. You must
enter at least one state.
 To override the pattern of a tone:

1. Select which tone you want to override in the Override Current Tone Values drop-down menu.
Figure 150: Tone Override Sections

1
2 3
4
5 6 7 8 9
• You can use the current values of the selected tone as a starting point for your
customization by clicking the Copy Current Tone Definition to Overridden button.
• You can clear all override fields by clicking the Reset Overridden Values button.
2. In the Overridden Tone Definition section, define the value of the proper Frequency used in the
corresponding Value field.
The value is in Hz. The range is from 10 Hz to 4000 Hz.
Note: You can use only two frequencies for the Call Waiting tone.
3. Define the power level of the proper Frequency in dBm in the corresponding Power field.
The range is from -99 dBm to 3 dBm.
4. If applicable, enter a value for the loop counter in the Loop Count field.
The range is from 2 to 128. This value will be used in Step 8.
Note: You can use only one loop count for the Call Waiting tone.
5. In the Overridden Tone States section, set the corresponding On/Off drop-down menu with the
proper value for each state.

Tone Override Software Configuration Guide
• On means the corresponding state plays a tone.

• Off means the corresponding state does not play a tone.
• CID means the moment where the Caller-ID will be sent to the analog port. This options
is available only for the Call Waiting tone.
You may also want to perform the following operations:
• To add a state, click the button at the bottom of the Overridden Tone States section.
• To remove a state, click the button at the bottom of the Overridden Tone States
section. This removes the last state in the list.
6. For the On states, select the frequency to play in the corresponding Frequencies column.
The frequencies defined in the Overridden Tone Definition section are listed as clickable buttons.
You can use from one to four frequencies. A blue button indicates that the frequency is selected.
7. Set the corresponding Duration field with the number of times, in ms, to perform the action of the
state.
The range is from 10 ms to 56000 ms. The tone stays indefinitely in the state (continuous) if no time
is specified.
8. In the corresponding Loop drop-down menu, select whether or not to stop looping between states
after a number of loops defined in Step 4.
When the number of loops is reached, the next state is s(n+1) for the state s(n) instead of the state
defined in the Next State drop-down menu.
9. In the corresponding Next State drop-down menu, select the next tone state to use when the time
has elapsed.
This value is not available if the Duration field is empty.

Chapter 35 - Tone Customization Parameters Configuration Tone Override

C H A P T E R
36 Music on Hold Parameters

Configuration
This chapter describes how to configure the Music on Hold (MoH) parameters.
 MP3 file download server setup.
 Music on Hold configuration.
Standards Supported • RFC 1350: The TFTP Protocol (Revision 2) (client-side only)
• RFC 2616: Hypertext Transfer Protocol - HTTP/1.1 (client-
side only)
MP3 File Download Server
To download a MP3 file, you may need to setup the following applications on your computer:
 TFTP server with proper root path
 HTTP server with proper root path
Configuring the TFTP Server

When you perform a MP3 file download by using the TFTP (Trivial File Transfer Protocol) protocol, you must
install a TFTP server running on the PC designated as the TFTP server host. It is assumed that you know how
to set the TFTP root path. If not, refer to your TFTP server’s documentation.
Configuring the HTTP Server

When you to perform a MP3 file download by using the HTTP protocol, you must install a HTTP server running
on the PC designated as the server host. It is assumed that you know how to set the root path. If not, refer to
your HTTP server’s documentation.
Music on Hold Configuration
The Music on Hold sub-page of the Telephony page allows you to configure the music (in the form of an MP3
file) that plays when a local user has been put on hold. Note that transfers exceeding 5 minutes are cancelled.

Chapter 36 - Music on Hold Parameters Configuration Music on Hold Configuration
 To set the Music on Hold parameters:

1. In the web interface, click the Telephony link, then the Music on Hold sub-link.
Figure 151: Telephony – Music on Hold Web Page
3
4
5
2. In the Music On Hold Configuration section, indicate whether or not the unit should play music when
being put on hold in the Streaming drop-down menu.
When enabled, music is played toward the telephony side when being put on hold from the network
side.
3. In the Transfer Configuration section, enter the URL to the MP3 file to use in the URL field.
This file is loaded when the Aastra unit starts and reloaded every time the Reload Interval value
elapses (see Step 5). It must be smaller than 1024 Kilobytes unless otherwise specified in a
customer profile.
The MP3 file downloaded must be encoded with a sampling rate of 8000 Hz (only available through
MPEG version 2.5) and in mono channel mode. All other types of file will be rejected. The decoding
output will be in mono channel mode, with a sample rate of 8000 Hz and with 8 bits per sample.
You can use the following supported protocols to transfer the file:
• HTTP: HyperText Transfer Protocol.
• TFTP: Trivial File Transfer Protocol.
URLs using any other transfer protocol are invalid.
Note: The HTTP protocol does not support spaces between characters in the URL.
Examples of valid URLS:

• http://www.myserver.com/myfile.mp3
• tftp://myserver.com:69/myfolder/myfile.mp3
When the port is not included in the URL, the default port for the chosen protocol is used.
HTTP supports basic or digest authentication mode as described in RFC 2617.
If you have selected HTTP, please note that your server may activate some caching mechanism for
the MP3 download. This mechanism caches the initial MP3 download for later processing, thus
preventing changes of the original MP3.
4. If your server requires authentication when downloading the MP3, set the following:
• The user name in the User Name field.
• The password in the Password field.

Music on Hold Configuration Software Configuration Guide
5. Set the time, in hours, between attempts to load the MP3 file in the Reload Interval field.
If you enter the value 0, this means that the unit loads the file only once at unit startup. Any other
value between 1 and 6000 is the number of hours between automatic reloads of the file. When a
manual file download is triggered, the counter is not reset so the next reload will happen at the same
time.
• To save your settings without transferring the MP3 file, click Submit.
• To save your settings and transfer the MP3 file now, click Submit & Transfer Now.
• To save your settings and stop a file transfer in progress, click Submit & Cancel
Transfer.

Chapter 36 - Music on Hold Parameters Configuration Music on Hold Configuration

C H A P T E R
37 Country Parameters
Configuration
This chapter describes how to configure the country information:

 Select a specific country.
 Additional country settings.
 Call Detail Record
Country Configuration
The Misc sub-page of the Telephony page allows you to configure the country in which the unit is located.
 To set the miscellaneous parameters:

1. In the web interface, click the Telephony link, then the Misc sub-link.
Figure 152: Telephony – Misc Web Page
2. In the Country section, select the country in which the Aastra unit is located in the Country Selection
drop-down menu.
It is very important to set the country in which the unit is used because a number of parameter
values are set according to this choice, such as tones, rings, impedances, and line attenuations.
See “Appendix A - Country-Specific Parameters” on page 603 for more information on these
country-specific settings.

Chapter 37 - Country Parameters Configuration Additional Country Settings
Additional Country Settings

Input/Output User Gain

The user gain allows you to modify the input and output sound level of the Aastra unit.
Caution: Use these settings with great care. Aastra recommends not to modify the user gain variables
unless absolutely necessary because default calibrations may no longer be valid.
Modifying user gains may cause problems with DTMF detection and voice quality – using a high user gain
may cause sound saturation (the sound is distorted). Furthermore, some fax or modem tones may no longer
be recognized. The user gains directly affect the fax communication quality and may even prevent a fax to
be sent.
You can compensate with the user gain if there is no available configuration for the country in which the Aastra
unit is located. Because the user gain is in dB, you can easily adjust the loss plan, e.g., if you need an
additional 1 dB for analog to digital, put 1 for user gain output.
You can use two types of configuration as described in “Default vs. Specific Configurations” on page 326.
 To set user gain variables:

1. In the telIfMIB, locate the countryCustomizationUserGainGroup folder.
2. Define the default user output gain offset in dB (from analog to digital) in the
defaultCountryCustomizationUserGainOutputOffset variable.
telIf.defaultCountryCustomizationUserGainOutputOffset="Value"
Values range from -12 dB to +12 dB. However, going above +6 dB may introduce clipping/distortion
depending on the country selected.
3. If you want to set a different output gain offset for one or more interfaces, set the following variables:
• specificCountryCustomizationUserGainEnableConfig variable for the specific
interface you want to configure to enable.
• specificCountryCustomizationUserGainOutputOffset variable for the specific line
you want to configure.
telIf.specificCountryCustomizationUserGain.EnableConfig[InterfaceId="Interface"]
="1"
telIf.specificCountryCustomizationUserGain.OutputOffset[InterfaceId="Interface"]

Additional Country Settings Software Configuration Guide
="Value"
where:
• Interface is the name of the interface you want to configure (for instance, Slot2/Pri1).
• Value is the output gain offset.
4. Define the default user input gain offset in dB (from digital to analog) in the
defaultCountryCustomizationUserGainInputOffset variable.
telIf.defaultCountryCustomizationUserGainInputOffset="Value"
Values range from -12 dB to +12 dB. However, going above +6 dB may introduce clipping/distortion
depending on the country selected.
5. If you want to set a different input gain offset for one or more interfaces, set the following variables:
• specificCountryCustomizationUserGainEnableConfig variable for the specific
• specificCountryCustomizationUserGainInputOffset variable for the specific line
you want to configure.
telIf.specificCountryCustomizationUserGain.EnableConfig[InterfaceId="Interface"]
="1"
telIf.specificCountryCustomizationUserGain.InputOffset[InterfaceId="Interface"]=
"Value"
where:
• Interface is the name of the interface you want to configure (for instance, Slot2/Pri1).
• Value is the input gain offset.
6. Restart the TelIf service by accessing the scmMIB and setting the serviceCommandsRestart
variable for the TelIf service to restart.
scm.serviceCommands.Restart[Name=TelIf]="10"
Dialing Settings
Dialing settings allow you to configure how the Aastra unit dials numbers.
When selecting a country (see “Country Configuration” on page 325 for more details), each country has default
dialing settings. However, you can override these values and define your own dialing settings.
 To set the dialing settings:

1. In the telIfMIB, locate the countryCustomizationDialingGroup folder.
2. Set the defaultCountryCustomizationDialingOverride variable to enable.
telIf.specificCountryCustomizationDialing.EnableConfig[InterfaceId="Interface"]=
"1"
where Interface is the name of the interface you want to configure (for instance, Slot2/Pri1).
This allows overriding the default country settings.
3. If you want to change the override status for one or more interfaces, set the following variables:
• specificCountryCustomizationDialingEnableConfig variable for the specific
• specificCountryCustomizationDialingOverride variable for the specific interface
you want to configure to enable.

Chapter 37 - Country Parameters Configuration Additional Country Settings
"1"
telIf.specificCountryCustomizationDialing.Override[InterfaceId="Interface"]="1"
4. Set an inter-digit dial delay in the defaultCountryCustomizationDialingInterDtmfDialDelay
variable.
telIf.defaultCountryCustomizationDialing.InterDtmfDialDelay="Value"
This is the delay, in milliseconds (ms), between two DTMFs when dialing the destination phone
number. Values range from 50 ms to 600 ms.
5. If you want to set a different inter-digit dial delay for one or more interfaces, set the following
variables:
• specificCountryCustomizationDialingInterDtmfDialDelay variable for the
specific interface you want to configure.
"1"
telIf.specificCountryCustomizationDialing.InterDtmfDialDelay[InterfaceId="Slot3/
Bri3"]="Value"
6. Set the DTMF duration value in the defaultCountryCustomizationDialingDtmfDuration
variable.
telIf.defaultCountryCustomizationDialing.DtmfDuration="Value"
This is the duration, in milliseconds (ms), a DTMF is played when dialing the destination phone
number. Values range from 50 ms to 600 ms.
7. If you want to set a different DTMF duration value for one or more interfaces, set the following
variables:
• specificCountryCustomizationDialingDtmfDuration variable for the specific
interface you want to configure.
"1"
telIf.specificCountryCustomizationDialing.DtmfDuration[InterfaceId="Interface"]=
"Value"
8. Set the delay, in milliseconds, between two MFR1s when dialing on the interface in the
DefaultCountryCustomizationDialingInterMfR1DialDelay variable.
See “Chapter 23 - E&M CAS Configuration” on page 253 for more details on MFR1 signalling.
9. Set the delay, in milliseconds, between two MFR1s when dialing on the interface by putting the
following line in the configuration script:
telIf.defaultCountryCustomizationDialing.InterMfR1DialDelay="Value"
Values range from 50 ms to 600 ms.
10. If you want to set a different delay value for one or more interfaces, set the following variables:

Additional Country Settings Software Configuration Guide
"1"
telIf.specificCountryCustomizationDialing.InterMfR1DialDelay[InterfaceId="Interf
ace"]="Value"
11. Set the duration, in milliseconds, of a MFR1 when dialling on the interface in the
DefaultCountryCustomizationDialingMfR1Duration variable.
See “Chapter 23 - E&M CAS Configuration” on page 253 for more details on MFR1 signalling.
12. Set the duration, in milliseconds, of a MFR1 when dialing on the interface by putting the following
line in the configuration script:
telIf.DefaultCountryCustomizationDialing.MfR1Duration="Value"
Values range from 50 ms to 600 ms.
13. If you want to set a different duration value for one or more interfaces, set the following variables:
"1"
telIf.specificCountryCustomizationDialing.MfR1Duration[InterfaceId="Interface"]=
"Value"
14. Restart the TelIf service by accessing the scmMIB and setting the serviceCommandsRestart
variable for the TelIf service to restart.
scm.serviceCommands.Restart[Name=TelIf]="10"
Fax Calling Tone Detection

You can enable the fax calling tone (CNG tone) detection.
 To enable fax calling tone detection:

1. In the telIfMIB, locate the machineDetectionGroup folder.
2. Set the defaultMachineDetectionCngToneDetection variable to enable.
telIf.defaultMachineDetection.CngToneDetection="1"
Upon recognition of the CNG tone, the Aastra unit switches the communication from voice mode to
fax mode and the CNG is transferred by using the preferred fax codec. This option allows for quicker
fax detection, but it also increases the risk of false detection.
If you do not want the Aastra unit to detect the fax calling tone, set the variable to disable(0). In this
case, the CNG tone does not trigger a transition from voice to data and the CNG is transferred in
the voice channel. With this option, faxes are detected later, but the risk of false detection is
reduced.
3. If you want to set a different calling tone detection setting for one or more interfaces, set the
following variables:
• specificMachineDetectionEnableConfig variable for the specific interface you want
to configure to enable.
• specificMachineDetectionCngToneDetection variable for the specific interface you
want to configure.
telIf.specificMachineDetection.EnableConfig[InterfaceId="Interface"]="1"
telIf.specificMachineDetection.CngToneDetection[InterfaceId="Interface"]="Value"

Chapter 37 - Country Parameters Configuration CDR (Call Detail Record)
CDR (Call Detail Record)
Call detail record (CDR) in VoIP contains information about recent system usage such as the identities of
sources (points of origin), the identities of destinations (endpoints), the duration of each call, the total usage
time in the billing period and many others.
The Misc sub-page of the Telephony page allows you to configure the CDR parameters.
 To set the CDR parameters:

1. In the Call Detail Record section of the Misc page, set the host name and port number of the device
that archives CDR log entries in the Syslog Remote Host field.
Specifying no port (or port 0) sends notifications to port 514.
Figure 153: CDR Call Detail Record Section
1
2
3
2. Specify the format of the syslog Call Detail Record in the Syslog Format field.
The formal syntax description of the protocol is as follows:
Precision=DIGIT
Width=DIGIT
MacroId=(ALPHA / "_")
Macro=%[Width]|[.Precision]|[Width.Precision]MacroId
The Width field is the minimum width of the converted argument. If the converted argument has
fewer characters than the specified field width, then it is padded with spaces. If the converted
argument has more characters than the specified field width, the field width is extended to whatever
is required.
The Precision field specifies the maximum number of characters to be printed from a string.
Examples :
sipid=SipUser001
CDR Log: %sipid --> CDR Log : SipUser001
CDR Log: %15sipid --> CDR Log : SipUser001
CDR Log: %15.5sipid --> CDR Log : SipUs
CDR Log: %.5sipid --> CDR Log : SipUs
Call Detail Record predefined macros.
Control characters:
Table 242: Control Character
Character Value
%% %
\n Split message
Call detail record macros:

Table 243: Call Detail Record Macros
Macro Value
%id CDR ID. The CDR ID is unique. The ID is incremented by one each time it is
represented in a CDR record
%sipid SIP call ID. Blank if no SIP interface was used during the call.
%ocgnum Original calling number. Calling number as received by the unit.
%cgnum Calling number. Calling number after manipulation by the call router.

CDR (Call Detail Record) Software Configuration Guide
Table 243: Call Detail Record Macros (Continued)
Macro Value
%ocdnum Original called number. Called number as received by the unit.

%cdnum Called number. Called number after manipulation by the call router.
%oiname Original Interface name. Interface on which the call was received. Ex. isdn-Slot2/
Pri1.
%diname Destination interface name. Interface on which the call was relayed. Ex. SIP-
Default
%chan Channel number. Blank if no PRI/BRI interface was used during the call. If 2 PRI/
BRI interface were involved, display the originating interface.
%sipla SIP local IP address.
%sipra SIP remote IP address or FQDN (next hop).
%siprp SIP remote port (next hop).
%mra Media remote IP address. Source IP address of incoming media stream. If the
stream was modified during the call, display the last stream.
%mrsp Media remote port. Source port of incoming media stream. If the stream was
modified during the call, display the last stream.
%mdrp Media remote port. Destination port of outgoing media stream. If the stream was
modified during the call, display the last stream.
%tz Local time zone
%cd Call duration (in seconds) (connect/disconnect).
%sd Call duration (in seconds) (setup/connect).
%pdd Post dial delay (in seconds) (setup/progress).
%css Call setup second (local time)
%csm Call setup minute (local time)
%csh Call setup hour (local time)
%csd Call setup day (local time)
%csmm Call setup month (local time)
%csy Call setup year (local time)
%ccs Call connect second (local time)
%ccm Call connect minute (local time)
%cch Call connect hour (local time)
%ccd Call connect day (local time)
%ccmm Call connect month (local time)
%ccy Call connect year (local time)
%cds Call disconnect second (local time)
%cdm Call disconnect minute (local time)
%cdh Call disconnect hour (local time)
%cdd Call disconnect day (local time)
%cdmm Call disconnect month (local time)

Chapter 37 - Country Parameters Configuration CDR (Call Detail Record)
Table 243: Call Detail Record Macros (Continued)
Macro Value
%cdy Call disconnect year (local time)

%miptxc IP Media last transmitted codec
%miptxp IP Media last transmitted p-time
%dr Disconnect reason (ISDN reason codes with ISUP SIP mapping)
%rxp Received media packets. Excluding T.38.
%txp Transmitted media packets. Excluding T.38.
%rxpl Received media packets lost. Excluding T.38.
%rxmd Received packets mean playout delay (ms, 2 decimals). Excluding T.38.
%rxaj Received packets average jitter (ms, 2 decimals). Excluding T.38.
%sipdr SIP disconnect or rejection reason.
3. Set the Syslog facility used by the unit to route the Call Detail Record messages in the Syslog
Facility field.
The application can use Local0 through Local7.

Call Router Parameters
C H A P T E R
38 Call Router Configuration
This chapter describes the call router service.

 Introduction to the call router’s parts and types supported.
 Routes parameters.
 Mappings parameters.
 Call signalling parameters.
 SIP headers translation parameters.
 Call properties translation parameters.
 Hunt table parameters.
 SIP Redirects parameters.
Standards Supported • ITU-T Recommendation E.164: The international public

telecommunication numbering plan.
• ITU-T Recommendation F.69: List of Telex Destination
Codes.
• ITU-T Recommendation X.121: International numbering plan
for public data networks.
Introduction
The Aastra unit’s call router allows you to route calls between interfaces. Based on a set of routing criteria, the
call router determines the destination (interface) for every incoming call. The forwarding decisions are based
on the following tables:
Table 244: Call Router Table Types
Table Description
Routing The routing table contains one or more routes. Each route associates a destination to a call
that matches a set of criteria. See “Routes” on page 353 for more details.
Mapping The mapping table contains one or more mapping types and expressions. A mapping
modifies call properties such as the calling and called party numbers according to the network
requirements. These mappings are specifically called within a route. See “Mappings” on
Call Call signalling specifies how to set up a call to the destination Aastra unit or 3rd party
Signalling equipment. Call signalling properties are assigned to a route and used to modify the
behaviour of the call at the SIP protocol level. See “Signalling Properties” on page 368 for
more details.
SIP A SIP headers translation overrides the default value of SIP headers in an outgoing SIP
Headers message. See “SIP Headers Translations” on page 372 for more details.
Translation
Call A call properties translation overrides the default value of call properties in an incoming SIP
Properties message. See “Call Properties Translations” on page 376 for more details.
Translation

Chapter 38 - Call Router Configuration Introduction
Table 244: Call Router Table Types (Continued)
Table Description
Hunt The hunt table contains one or more hunt entries, each with a set of possible destinations. A
hunt tries the destinations until one of the configured destinations accepts the call. See “Hunt
Service” on page 379 for more details.
SIP The SIP Redirects table allows configuring of SIP redirections that can be used as Route
Redirects destinations. When the Route source is a SIP interface, incoming SIP Invites are replied with
a 302 “Moved Temporarily” SIP response. See “SIP Redirects” on page 387 for more details.
When a new call comes from one of the Aastra unit interfaces, it is redirected to the routing table. The following
figure illustrates the Aastra unit call router:
Figure 154: Call Routing

Interfaces
Call Router
Route 1 Route 2 Route 3
Signalling Mapping 1
Properties 1
Hunt 1
Translation 1
(SIP Headers or
call properties)
Limitations
The call routing service has the following limitations:
 A call coming from a SIP interface cannot be routed to another SIP interface. When that occurs,
the call automatically fails.
 A call automatically fails if it is redirected to a route or hunt more than 10 times.
 Maximum number of Routes: 40
 Maximum number of Mapping Types: 40
 Maximum number of Mapping Expressions: 100
 Maximum number of Hunts: 40
 Maximum number of Signaling Properties: 40
 Maximum number of SIP Header Translations: 100
 Maxium number of Call Properties Translations: 100

Introduction Software Configuration Guide
Regular Expressions
Standards Supported • IEEE Std 1003.1-2001: IEEE Standard for Information
Technology---Portable Operating System Interface (POSIX®)
Some of the routing types described in “Routing Type” on page 338 require that you enter them following the
regular expression syntax. A regular expression is a string used to find and replace strings in other large
strings. The Aastra unit uses regular expressions to enter a value in several routing types, often by using
wildcard characters. These characters provide additional flexibility in designing call routing and decrease the
need for multiple entries in configuring number ranges.
The expression cannot begin by “^”, it is implicit in the expression. The following table shows some of the
wildcard characters that are supported:
Table 245: Regular Expressions Wildcards
Character Description
. Single-digit place holder. For instance, 555 .... matches any dialed number beginning with
555, plus at least four additional digits. Note that the number may be longer and still match.
* Repeats the previous digit 0, 1, or more times. For instance, in the pattern:
1888*1
the pattern matches:
1881, 18881, 188881, 1888881
Note: If you are trying to handle the asterisk (*) as part of a dialed number, you must use \*.
[] Range of digits.
• A consecutive range is indicated with a hyphen (-), for instance, [5-7].
• A nonconsecutive range is indicated without a delimiter, for instance, [58].
• Both can be used in combination, for instance [5-79], which is the same as
[5679].
You may place a (^) symbol right after the opening bracket to indicate that the specified range
is an exclude list. For instance, [^01] specifies the same range as [2-9].
Note: The call router only supports single-digit ranges. You cannot specify the range of
numbers between 99 and 102 by using [99-102].
() Indicates a pattern (also called group), for instance, 555(2525). It is used when replacing a
number in a mapping. See “Groups” on page 337 for more details.
? Matches 0 or 1 occurrence of the previous item. For instance, 123?4 matches both 124 and
1234.
+ Repeats the previous digit one or more time. For instance 12+345 matches 12345, 122345,
etc. (but not 1345). If you use the + at the end of a number, it repeats the last number one or
more times. For instance: 12345+ matches, 12345, 123455, 1234555, etc.
| Indicates a choice of matching expressions (OR).
The matching criterion implicitly matches from the beginning of the string, but not necessarily up to the end.
For instance, 123 will match the criterion 1, but it will not match the criterion 2.
If you want to match the whole string, you must end the criterion with “$”. For instance, 123 will not match the
criterion 1$ and will match the criterion 123$.
Note: You can use the “<undefined>” string if you want to match a property that is not defined.
You can also use the macro “local_ip_port“ to replace the properties by the local IP address and port of the
listening network of the SIP gateway used to send the INVITE.
Groups

A group is placed within parenthesis. It is used when replacing a string in a mapping. You can use up to nine
groups (defined by “\1” to “\9”) and matching is not case sensitive. “\0” represents the whole string. Lets say
for instance you have the following string:
9(123(45)6)
The following describes how the groups are replaced in a properties manipulation:
Table 246: Groups Replacement Example
Replacement Result
\0 9123456
\1 123456
\2 45
\3
Groups can only be used with the following routing types:

 Calling/Called E.164
 Calling/Called Name
 Calling/Called Host
 Calling/Called URI
Routing Type
Standards Supported • ITU-T Recommendation Q.931: ISDN user-network interface
layer 3 specification for basic call control
The following sub-sections list the available routing types of the call router and their supported values. The
routing types that offer choices use the choices as defined in the Q.931 standard. Q.931 is ISDN’s connection
control protocol, roughly comparable to TCP in the Internet protocol stack. The values may also be a special
tag, as described in “Special Tags” on page 344.
Table 247: Routing Types Locations
Routing Type Location
E164 “Called / Calling E164” on page 339

Type of Number (TON) “Called / Calling TON” on page 339
Numbering Plan Indicator (NPI) “Called / Calling NPI” on page 339
Name “Called / Calling Name” on page 340
Host “Called / Calling Host” on page 340
URI “Called / Calling URI” on page 340
Presentation Indicator (PI) “Calling PI” on page 340
Screening Indicator (SI) “Calling SI” on page 340
Information Transfer Capability (ITC) “Calling ITC” on page 341
Date and Time “Date/Time” on page 341
Phone Context “Called / Calling Phone Context” on page 342
SIP Username “Called / Calling SIP Username” on page 342
Bearer Channel “Called / Calling SIP Username” on page 342
Diverting Reason “Last / Original Diverting Reason” on page 342
Diverting E.164 “Last / Original Diverting E.164” on page 343

Table 247: Routing Types Locations (Continued)
Routing Type Location
Diverting Party Number Type “Last / Original Diverting Party Number Type” on page 343
Diverting Public Type Of Number “Last / Original Diverting Public Type Of Number” on page 343
Diverting Pivate Type Of Number “Last / Original Diverting Private Type Of Number” on page 343
Diverting Number Presentation “Last / Original Diverting Number Presentation” on page 344
SIP Privacy Type “SIP Privacy Type” on page 344
Aastra recommends to carefully define the routing requirements and restrictions that apply to your installation
before starting the routing configuration. This will help you determine the types of routing you need. When this
is done, define the routes and mappings, as well as the hunts that you need to fulfil these requirements. You
may need several entries of the same type to achieve your goals.
See also “Call Properties Parameters” on page 344 for a description of the parameters used by the various
routing types and interfaces of the call router.
Called / Calling E164

This is the Called/Calling Party Number. You can enter a regular expression (called/calling party E.164
number in the call setup message) as per “Regular Expressions” on page 337. Note that:
 A PBX may insert or modify the calling party number. Sometimes there is no calling party
number at all. This all depends on the equipment you connect to the device.
 The Aastra unit cannot filter the redirecting number information element of the SETUP
message because it does not support the “calling-Redir-E164” and “Calling-Redir-Reason”
routing properties criteria.
Called / Calling TON

Called or calling party type of number field in the ISDN setup message. The following values are available:
Table 248: Type of Number Values
Value Description
unknown Unknown number type.

international International number.
national National number.
network Network specific number used to indicate an administration or service number specific to the
serving network.
subscriber Subscriber number.
abbreviated Abbreviated number.
Note: The called type of number is set to international if the To username is an E.164 with the prefix “+”.
The calling type of number is set to international if the From username is an E.164 with the prefix “+”.
Called / Calling NPI

Called or calling party numbering plan indicator field in the ISDN setup message. The following values are

available:
Table 249: Numbering Plan Indicator Values
Value Description
unknown Unknown numbering plan.

isdn (E.164) ISDN/Telephony numbering plan according to ITU-T Recommendation E.164.
data (X.121) Data numbering plan according to ITU-T Recommendation X.121.
telex (F.69) Telex numbering plan according to ITU-T Recommendation F.69.
national Numbering plan according to a national standard.
private A private numbering plan.
Called / Calling Name

Calling and called party name (display name). This is the human-readable name of the calling or called party.
See “Regular Expressions” on page 337 for more details on how to enter a proper expression.
The Aastra unit does not support the sending of the calling name in the user-to-user information element.
Called / Calling Host

IP address or domain name of the called or calling host in the following format:
Fqdn[:port]
If [:port] is missing, the call router uses the well-known port of the signalling protocol. Note that:
 Incoming SIP calls use the calling party IP address property to store the IP address of the
remote SIP user agent. Other interfaces such as ISDN set the IP address to 0.0.0.0.
You can use a regular expression to enter an IP address or a range of IP addresses.
Called / Calling URI

Uniform Resource Identifier (URI) of:
 the called party, e.g., the To-URI.
 the originating VoIP peer, e.g., the From-URI of an incoming SIP call.
The URI follows the format described in RFC 3261.
Calling PI
Presentation indicator of the calling party number. The following values are available:
Table 250: Presentation Indicator Values
Value Description
allowed Presentation of the calling party number is allowed.

restricted Presentation of the calling party number is restricted.
interworking The calling party number is not available due to interworking.
You may want to remove the calling party number when the user sets the presentation indicator to restricted.
To achieve this, route restricted calls to a mapping that sets the Calling E164 to an empty string.
Calling SI

Screening indicator of the calling party number. The following values are available:
Table 251: Screening Indicator Values
Value Description
not- The user provides the calling party number but the number is not screened by the network.
screened Thus the calling party possibly sends a number that it does not own.
passed The calling party number is provided by the user and it passes screening.
failed The calling party number is set by the user and verification of the number failed.
network The originating network provides the number in the calling party number parameter.
You may want to remove the calling party number when it is not screened or screening failed. To do so, route
these calls to a mapping that sets the Calling E164 to an empty string. If you want to drop calls when the calling
party number is not screened or screening failed, use the Calling Si as criteria for the route.
Calling ITC
The information transfer capability field of the bearer capability information element in the ISDN setup
message. The following values are available:
Table 252: Information Transfer Capability Values
Value Description
speech Voice terminals (telephones).

unrestricted Unrestricted digital information (64 kbps).
restricted Restricted digital information (64 kbps).
3.1Khz Transparent 3.1 kHz audio channel.
udi-ta Unrestricted digital information with tones/announcements.
Note: This was formerly transparent 7.1 kHz audio channel.
video Video conference terminals.
The Aastra unit currently supports the following Information Transfer Capabilities when receiving calls to and
from the ISDN (named as in Q.931, 05/98):
 Speech
 Unrestricted Digital Information
 3.1 kHz Audio
Those are respectively referenced as Speech, Unrestricted and 3.1 kHz in the call routing configuration.
When initiating calls towards the ISDN, the Aastra unit uses the calling ITC value if it is one of the three listed
above. If none is set, it uses 3.1 kHz Audio. If the calling ITC set by the call router is different from the three
listed above, the call is rejected.
Note: Terminals connected to analog extensions (e.g. of a PBX) do not supply information transfer
capability values in their call setup. The configuration of the analog port on the Terminal Adapter, NT or PBX
is thus responsible to insert this value. The configuration of this value is however often omitted or wrong.
The ITC value may therefore not be a reliable indication to differentiate between analogue speech, audio or
Fax Group 3 connections. Furthermore, calls from SIP interfaces do not differentiate between bearer
capabilities. They always set the information transfer capability property to 3.1Khz.
Date/Time

Day of week and time period and/or date and time period. The following are the accepted formats:
Table 253: Date/Time Accepted Formats
Format Description
Date/Time Period format • 'DD.MM.YYYY/HH:MM:SS-DD.MM.YYYY/

HH:MM:SS'
• 'DD.MM.YYYY/HH:MM:SS-HH:MM:SS'
• 'DD.MM.YYYY-DD.MM.YYYY'
• 'DD.MM.YYYY'
• 'HH:MM:SS-HH:MM:SS'
Week Day/Time Period format • 'DDD'
• 'DDD,DDD...'
• 'DDD/HH:MM:SS-HH:MM:SS'
• 'DDD,DDD.../HH:MM:SS-HH:MM:SS'
DDD must be one of: SUN, MON, TUE, WED, THU, FRI,
SAT.
Many of the formats above can be concatenated to form one expression. They must be separated by |. For
instance: 25.12.2006 | SUN.
Called / Calling Phone Context

This is a user parameter in a URI. For instance:
sip:1234;[email protected];user=phone
You can enter a regular expression (called/calling party phone context in the call setup message) as per
“Regular Expressions” on page 337.
Called / Calling SIP Username

Calling and called party SIP username. See “Regular Expressions” on page 337 for more details on how to
enter a proper expression.
Called / Calling Bearer Channel

Calling and called party bearer channel. See “Regular Expressions” on page 337 for more details on how to
enter a proper expression.
Last / Original Diverting Reason

Standards Supported • RFC 5806: Diversion Indication in SIP
This is the last or original diverting reason in ISDN setup and SIP INVITE messages. The following values are
available:
Table 254: Diverting Reason Values
Value Description
cfb Call Forward on Busy – Allowed.

cfu Call Forward on Unavailable – Restricted
cfnr Call Forward on No Answer – Interworking
unknown unknown

Refer to “You can set the SIP transfer method when an endpoint is acting as the transferor in a blind transfer
scenario.” on page 345 to select the SIP method used to receive/send call diversion information in an INVITE.
Last / Original Diverting E.164

Last or original party number to which the call was being routed when the first diversion occurred. You can
enter a regular expression (called/calling party E.164 number in the call setup message) as per “Regular
Expressions” on page 337. Note that:
 A PBX may insert or modify the calling party number. Sometimes there is no calling party
number at all. This all depends on the equipment you connect to the device.
 The Aastra unit cannot filter the redirecting number information element of the SETUP
message because it does not support the “calling-Redir-E164” and “Calling-Redir-Reason”
routing properties criteria.
Last / Original Diverting Party Number Type

Table 255: Diverting Party Number Type Values
Value Description

public Public number.
private Private number.
Last / Original Diverting Public Type Of Number

Diverting or original called number public type of number field in the ISDN Setup message. Used only when
the diverting or original called number type of number is 'public'. The following values are available:
Table 256: Diverting Public Type of Number Values
Value Description

international International number.
national National number.
network- Network specific number used to indicate an administration or service number specific to the
specific serving network.
Last / Original Diverting Private Type Of Number

Diverting or original called number private type of number field in the ISDN Setup message. Used when the
diverting or original called party number type is 'private'. The following values are available:
Table 257: Diverting Private Type of Number Values
Value Description
unknown Unknown.
leg2-reg Leg2 reg.
leg1-reg Leg1 reg.

Table 257: Diverting Private Type of Number Values (Continued)
Value Description
pisn- PISN Specific.

specific
Last / Original Diverting Number Presentation

Diverting or original called number presentation. The following values are available:
Table 258: Diverting Presentation Values
Value Description
allowed Presentation of the party number is allowed.

restricted Presentation of the party number is restricted.
interworking The party number is not available due to interworking.
restricted- Restricted address.
address
SIP Privacy Type

Calling SIP privacy level of the call. The following values are available:
Table 259: SIP Privacy Values
Value Description
disabled No privacy is used.

none Use P-Asserted Identity privacy.
id Use P-Preferred Identity privacy.
Special Tags
You can use the following special tags as routing types values.
Table 260: Special Tags
Tag Description
undefined Matches if the property is not defined for the call.

default Always matches. Generally used to set a default route if the previous criteria do not match.
Call Properties Parameters

The following sections describe the parameters used by the various call properties (routing types) and
interfaces of the call router.
Call Properties to SIP

This section describes the information the call router uses for the various SIP fields.
Table 261: Call Properties to SIP
SIP Field Description
To The Aastra unit uses the calling URI to populate the To field if not undefined.
Otherwise, the unit does the following:
• Uses the called Name for the friendly name if not undefined.
• Uses the called SipUsername for the user name if not empty or
undefined; otherwise, uses the called E164 for the username. If it is
empty or undefined, the Aastra unit rather uses the value defined in
the Default Username Value field of the SIP > Interop > SIP Interop
parameters as username (see “SIP Interop” on page 312 for more
details). The unit uses the called Phone Context for the user's
'phone-context' parameter if not empty. If a 'phone-context'
parameter is added, the URI parameter 'user' is also automatically
added. Its value is defined in the SIP URI User Parameter Value
field of the SIP > Interop > SIP Interop parameters. If empty, then
the value 'phone' is used
• Uses the called Host for the host if not undefined, otherwise uses the
configured home domain proxy host.
• Prefixes the user name with “+” and adds the URI parameter “user”
with the value “phone” if the called TON is “international”.
• If there is no URI parameter “user” yet and the SIP URI User
Parameter Value field of the SIP > Interop > SIP Interop parameters
is not empty, then the parameter is added with the value defined by
the field.
From The Aastra unit uses the called URI to populate the From field if not undefined.
Otherwise, the unit does the following:
• Uses the calling Name for the friendly name if not undefined.
• Uses the calling SipUsername for the user name if not empty or
undefined; otherwise, uses the calling E164 for the username. If it is
empty or undefined, the Aastra unit rather uses the value defined in
the Default Username Value field of the SIP > Interop > SIP Interop
parameters as username (see “SIP Interop” on page 312 for more
details).The unit uses the calling Phone Context for the user's
'phone-context' parameter if not empty. If a 'phone-context'
parameter is added, the URI parameter 'user' is also automatically
added. Its value is defined in the SIP URI User Parameter Value
field of the SIP > Interop > SIP Interop parameters. If empty, then
the value 'phone' is used.
• Uses the calling Host for the host if not undefined, otherwise uses
the configured home domain proxy host.
• Prefixes the user name with “+” and adds the URI parameter “user”
with the value “phone” if the calling TON is “international”.
• If there is no URI parameter “user” yet and the SIP URI User
Parameter Value field of the SIP > Interop > SIP Interop parameters
is not empty, then the parameter is added with the value defined by
the field.
Request URI The Aastra unit uses the same information as the To field.
Contact The Aastra unit uses the same information as the From field, but with the
current IP address/port for the host.

Table 261: Call Properties to SIP (Continued)
SIP Field Description
Diversion A Diversion header is added if the Last Diverting E.164 property is present and
not empty. This Diversion header is constructed as follows:
• The username of the URI is set to the value of the Last Diverting
E.164 property.
• The host of the URI is set to the configured home domain proxy host.
• The reason field is set according to value of the Last Diverting
Reason property:
• cfu: "unconditional"
• cfb: "user-busy"
• cfnr: "no-answer"
• All other values or when undefined: "unknown'.
• The field counter is set to the value of DivertingCounter if the
Original Diverting E.164 property is set to empty or undefined,
otherwise it is set to DivertingCounter -1.
A second Diversion header is added if the Last Diverting E.164 and Original
Diverting E.164 properties are present and not empty. This Diversion header is
constructed as follows:
• The username of the URI is set to the value of the Original Diverting
E.164 property.
• The host of the URI is set to the configured home domain proxy host.
• The reason field is set according to the value of the Original
Diverting Reason property:
• cfu: "unconditional"
• cfb: "user-busy"
• cfnr: "no-answer"
• All other values or when undefined: "unknown'.
The field counter is set to 1.
SIP to Call Properties

This section describes the SIP information the call router uses for the various call properties.
Table 262: SIP to Call Properties
Property SIP Information
Called URI The URL of the To field.

Calling URI The URL of the From field.
Called Name The friendly name in the To field. The property is undefined if there is no
friendly name.
Calling Name The friendly name in the From field. The property is undefined if there is no
friendly name.
Called E164 The user name of the Request-Uri field if the user name is a compatible E.164.
The prefix “+” and separator “-” are removed. The property is undefined if there
is no user name or if it is not compatible.
Calling E164 The user name of the From field if the user name is a compatible E.164. The
prefix “+” and separator “-” are removed. The property is undefined if there is no
user name or if it is not compatible.
Called Host The host of the To field.

Table 262: SIP to Call Properties (Continued)
Property SIP Information
Calling Host The host of the Contact field.

Called TON Set to “international” if the To user name is an E.164 with the prefix “+”;
otherwise, the property is undefined.
Calling TON Set to “international” if the From user name is an E.164 with the prefix “+”;
otherwise the property is undefined.
Called Phone Context Set to the parameter “phone-context” of the user name of the To if the user
name is an E.164, otherwise the property is undefined.
Calling Phone Context Set to the parameter “phone-context” of the user name of the From if the user
name is an E.164, otherwise the property is undefined.
Called SIP Username Set to the username of the Request-Uri. Note that this does not include the
username parameter like the “phone-context”.
Calling SIP Username Set to the username of the From. Note that this does not include the username
parameter like the “phone-context”.
Last Diverting Reason If the INVITE contains at least one Diversion header, this value is set according
to the reason field value of the first Diversion header:
• "user-busy": cfb
• "unconditional":cfu
• "no-answer": cfna
• All other values: unknown
Otherwise, the property is undefined.
The reason field comparison is not case sensitive.
Original Diverting If the INVITE contains more than one Diversion header, this value is set
Reason according to the reason field value of the last Diversion header:
• "user-busy": cfb
• "unconditional":cfu
• "no-answer": cfna
• All other values: unknown
The reason field comparison is not case sensitive.
Last Diverting E.164 If the INVITE contains at least one Diversion header, this value is set to the
username of the URI (can be a SIP URI, SIPS URI or TEL URI) of the first
Diversion header converted into an E.164. It can be set to empty if there is no
username or if the username is not an E.164.
Original Diverting E.164 If the INVITE contains more than one Diversion header, this value is set to the
username of the URI (can be a SIP URI, SIPS URI or TEL URI) of the last
Diversion header converted into an E.164. It can be set to empty if there is no
username or if the username is not an E.164.
Diverting Counter If the INVITE contains at least one Diversion header, this value is set to the
sum of the counter field of all Diversion headers. If a diversion header does not
contain the counter field, the value 1 is assumed for the header.
All others The property is undefined.
Call Properties to ISDN

This section describes the information the call router uses for the various ISDN information elements.
Table 263: Call Properties to ISDN
Information Element Description
Bearer Capabilities If valid, the calling ITC is used to fill the “information transfer capability” (octet 3
[5:1]). Otherwise, the ITC is set to “3.1 kHz audio”. If more than one bearer
capability information elements is provided in a prioritized list, they all receive
the same ITC. This information element is included in the SETUP message
only for outgoing calls.
Calling Party Number Uses the calling E164 to fill the field “number digits” (octet 4).
Uses the calling TON to fill the field “type of number” (octet 3 [7:5]).
Uses the calling PI to fill the field “presentation indicator” (octet 3a [7:6).
Uses the calling SI to fill the field “screening indicator” (octet 3a [2:1]).
Uses the calling NPI to fill the field “numbering plan identification” (octet 3
[4:1]).
Called Party Number Uses the called E164 to fill the field “number digits” (octet 4).
Uses the called TON to fill the field “type of number” (octet 3 [7:5]).
Uses the called NPI to fill the field “numbering plan identification” (octet 3 [4:1]).
Display Uses the calling E164 to fill the field “display information” (octet 3).
Called Bearer Channel The called bearer channel is used to select a specific ISDN bearer channel for
an outgoing ISDN call.
ISDN to Call Properties

This section describes the ISDN information the call router uses for the various call properties.
Table 264: ISDN to Call Properties
Property ISDN Information
Calling Name Field “display information” (octet 3) of the Display information element, if
included in the SETUP Q.931 message.
Called E164 Field “number digits” (octet 4) of the called party information element included
in the SETUP Q.931 message.
Calling E164 Field “number digits” (octet 4) of the calling party information element included
in the SETUP Q.931 message.
Called TON Field “type of number” (octet 3 [7:5]) of the called party information element
Calling TON Field “type of number” (octet 3 [7:5]) of the calling party information element
Calling PI Field “presentation indicator” (octet 3a [7:6) of the calling party information
element included in the SETUP Q.931 message.
Calling SI Field “screening indicator” (octet 3a [2:1]) of the calling party information
element included in the SETUP Q.931 message.
Calling ITC Field “information transfer capability” (octet 3 [5:1]) of the bearer capability
information element included in the SETUP Q.931 message.
Called NPI Field “numbering plan identification” (octet 3 [4:1]) of the called party
Calling NPI Field “numbering plan identification” (octet 3 [4:1]) of the calling party

Table 264: ISDN to Call Properties (Continued)
Property ISDN Information
Calling Bearer Channel Represents the ISDN bearer channel on which the ISDN call is received.
All others The property is undefined.
Call Properties to FXS

This section describes the information the call router uses for the various call properties to FXS.
Table 265: Call Properties to FXS
Caller ID Description
Number If the PI property is present and not set to "allowed", the number is "P".
Otherwise, the number is set to the value of the E164 property (truncated to the
first 20 characters). See “Auto-Routing” on page 517 for details.
Name If the PI property is present and not set to "allowed", the name is "Anonymous".
Otherwise, the name is set to the value of the Name property (truncated to the
first 50 characters). See “Auto-Routing” on page 517 for details.
FXS to Call Properties

This section describes the information the call router uses for the various FXS to call properties.
Table 266: FXS to Call Properties
Calling E164 If the auto routing is enabled and the E164 field of the Call Router > Auto-
routing page is not empty (see “Auto-Routing” on page 517 for details), the
value of the E164 field. Otherwise, the property is not present.
Calling Name If the auto routing is enabled and the Name field of the Call Router > Auto-
value of the Name field. Otherwise, the property is not present.
Calling SIP Username If the auto routing is enabled and the SIP Username field of the Call Router >
Auto-routing page is not empty (see “Auto-Routing” on page 517 for details),
the value of the SIP Username field. Otherwise, the property is not present.
Called E164 For automatic calls, the E.164 defined in the Automatic Call Target field of the
Telephony > Services page (see “Automatic Call” on page 419 for more
details).
For other calls, the dialed digit after the transformation defined in the
Transformation field of the Allowed DTMF Map section (Telephony > DTMF
Maps page – see “Allowed DTMF Maps” on page 405 for more details).
Called Name For automatic calls, the name specified in the Automatic Call Target field of the
details). The property is not present if the target address does not contain a
name.
For other calls, the property is not present.

Table 266: FXS to Call Properties (Continued)
Called Host For automatic calls, the host specified in the the Automatic Call Target field of
the Telephony > Services page (see “Automatic Call” on page 419 for more
host.
For other calls, the host defined in the Target field of the Allowed DTMF Map
section (Telephony > DTMF Maps page – see “Allowed DTMF Maps” on
page 405 for more details). The property is not present if the target host is not
configured for the matching DTMF map.
Call Properties to FXO

This section describes the information the call router uses for the various call properties to FXO.
Table 267: Call Properties to FXO
Dialled number The Called E164 property.
FXO to Call Properties

This section describes the information the call router uses for the various FXO to call properties.
Table 268: FXO to Call Properties
Calling E164 If the caller ID is detected, the numbers provided by the caller ID.
If the auto routing is enabled and the E164 field of the Call Router > Auto-
value of the E164 field. Otherwise, the property is not present.
Calling Name If the caller ID is detected, the name provided by the caller ID.
If the auto routing is enabled and the Name field of the Call Router > Auto-
value of the Name field. Otherwise, the property is not present.
Calling SIP Username If the caller ID is detected, the property is not present.
If the auto routing is enabled and the SIP Username field of the Call Router >
Auto-routing page is not empty (see “Auto-Routing” on page 517 for details),
the value of the SIP Username field. Otherwise, the property is not present.
Called E164 For automatic calls, the E.164 defined in the Automatic Call Target field of the
details).
For other calls, the dialed digit after the transformation defined in the
Transformation field of the Allowed DTMF Map section (Telephony > DTMF
Maps page – see “Allowed DTMF Maps” on page 405 for more details).
Called Name For automatic calls, the name specified in the Automatic Call Target field of the
name.
For other calls, the property is not present.

Table 268: FXO to Call Properties (Continued)
Called Host For automatic calls, the host specified in the the Automatic Call Target field of
the Telephony > Services page (see “Automatic Call” on page 419 for more
host.
For other calls, the host defined in the Target field of the Allowed DTMF Map
section (Telephony > DTMF Maps page – see “Allowed DTMF Maps” on
page 405 for more details). The property is not present if the target host is not
configured for the matching DTMF map.
SIP/ISDN Call Default Values

When performing a call from SIP to ISDN or ISDN to SIP, some ISDN informations are missing from the SIP
packet. The Dgw v2.0 Application sets the following default values when the information is missing. You
cannot filter on these default values, but you can filter with the “<undefined>” or “<default>” values.
Table 269: SIP/ISDN Calls Default Values
Parameter Default Value
SIP to ISDN Calls

TON (calling) unknown
TON (called) unknown
NPI (calling and called) unknown
SI (calling) User-side: not-screened
Network-side: network
ITC (calling) 3.1 kHz audio
PI (calling) 1. When the Calling Party Number E.164 is missing: interworking. In this
case, this value overrides any value set by the call router.
2. When CLIR is enabled (user-side only): restricted. In this case, this value
overrides any value set by the call router.
3. All other cases: allowed. This is the default value if the two cases above
do not apply and no value has been set by the call router.
ISDN to SIP Calls
SI (calling) Network-side: The SI in the incoming Calling Party information element is ignored
and replaced by one of the following:
1. No calling IA5 digits received: network.

2. NPI is not “unknown” nor “ISDN telephony”: network.
3. TON is not “international” nor “national”: network, called IA5 digits are
discarded.
4. PI is set to “interworking”: network.
5. Otherwise: passed.
User-side: not-screened.

Table 269: SIP/ISDN Calls Default Values (Continued)
Parameter Default Value
PI (calling) Network-side:
1. CLIR enabled: restricted. The PI is set to restricted no matter if a PI is

present in the incoming Calling Party IE.
2. CLIR disabled, no IA5 digits provided: interworking.
3. CLIR disabled, IA5 digits provided: allowed.
User-side:
1. CLIR disabled, no IA5 digits provided: interworking.

2. CLIR disabled, IA5 digits provided: allowed.
ITC (calling) Must be provided in the incoming Bearer Capabilities information element
provided by the ISDN peer that initiated the call. There is no default value, the call
should be rejected if missing.
TON (called) The Called TON must be provided by the ISDN peer that initiated the call.
TON (calling) unknown
NPI (called The Called NPI must be provided by the ISDN peer that initiated the call.
NPI (calling) unknown
Note that the calling PI, SI, TON and NPI are present in Calling Party information elements in SETUP
messages sent by the network-side only when CLIP is enabled. They should always be present in messages
sent by the user-side. See “Chapter 21 - ISDN Configuration” on page 177 for more details on CLIP.
Call Routing Status

The routes, mappings, and hunts currently in use, as well as the available interfaces, are displayed in the Call
Router > Status page.

Routes Software Configuration Guide
Figure 155: Call Router – Status Web Page
Routes
The routing table contains one or more routes. These routes forward an incoming or outgoing call to another
route, interface, or hunt based on a specific call property such as the called party number. It may also use a
mapping to modify the call setup message of a call and a signalling property to modify the behaviour of the
call at the SIP protocol level.
Once the call router finds a route that matches, it does not check the other routes, even if some of them may
still match. The routes sequence is thus very important. The call router follows the routing table rows (routes)
as they are entered in the web interface. If you want the call router to try to match one row before another one,
you must put that row first.
When a call arrives, the call router proceeds as follows:
1. It examines the call property as specified with the routes.
To select a route, the call must match all three of the Source, Properties Criteria, and Expression
Criteria parameters.
2. It selects the first matching route in the list of routes.

Chapter 38 - Call Router Configuration Routes
3. It routes the call to the specified destination interface, hunt, or route.
Note: You can revert back to the configuration displayed in the Call Router > Status web page at any time
by clicking the Rollback button at the bottom of the page. All modified settings in the Call Router > Route
Config page will be lost.
You can add up to 40 routes.
Creating/Editing a Route
The web interface allows you to create a route or modify the parameters of an existing one.
 To create or edit a route:

1. In the web interface, click the Call Router link, then the Route Config sub-link.
Figure 156: Call Router – Route Config Web Page
2. Locate the Route section.

• If you want to add a route before an existing entry, locate the proper row in the table
and click the button of this row.
• If you want to add a route at the end of the existing rows, click the button at the
bottom right of the Route section.
• If you want to edit an existing route, locate the proper row in the table and click the
button.
This brings you to the Configure Route panel.
Figure 157: Configure Route Panel
4
5
6
7
8
9

4. Enter one or more sources to compare with the call and match in order to select the route in the
Source field.
A source may be:
• route-name: The call uses the route name.
• sip-name: The call comes from the SIP interface name.
• isdn-name: The call comes from the ISDN interface name.
• r2-name: The call destination is set to the R2 interface name.
• e&m-name: The call comes from the E&M interface name.
• fxs-name: The call destination is set to the FXS interface name.
• fxo-name: The call destination is set to the FXO interface name.
If you want to use multiple sources, you must separate them by commas.
For instance, if you want to route calls that come from the SIP interface “default”, enter the following
value:
sip-default
If you want to route calls that come from the SIP interfaces “default” and “other”, enter the following
value:
sip-default,sip-other
Keep in mind that to select a route, the call must match all three of the Source, Properties Criteria,
and Expression Criteria parameters.
5. Select a call property to compare with the call and match in order to select the route in the Properties
Criteria drop-down menu.
The call router offers several different routing types. Each type specifies which call property the call
router examines.
Table 270: Routing Types
Type Description
Called E164 Routes calls based on the called party E.164 number.
Calling E164 Routes calls based on the calling party E.164 number.
Called TON Routes calls based on the called party type of number.
Calling TON Routes calls based on the calling party type of number.
Called NPI Routes calls based on the called party numbering plan indicator.
Calling NPI Routes calls based on the calling party numbering plan indicator.
Called Name Routes calls based on the display name of the called party.
Calling Name Routes calls based on the display name of the calling party.
Called Host Routes calls based on the signalling IP address or domain name.
Calling Host Routes calls based on the signalling IP address or domain name.
Called URI Routes calls based on the To-URI.
Calling URI Routes calls based on the From-URI.
Calling PI Routes calls based on the presentation indicator.
Calling SI Routes calls based on the screening indicator.
Calling ITC Routes calls based on the information transfer capability.
Date/Time Routes calls based on the date and/or time the call arrived at the call router. A
link called Time criteria editor appears on the right of the Expression criteria
field. Use it to easily configure the Date/Time type.

Chapter 38 - Call Router Configuration Routes
Table 270: Routing Types (Continued)
Type Description
Called Phone Routes calls based on the called party phone context.
Context
Calling Routes calls based on the calling party phone context.
Phone
Context
Called SIP Routes calls based on the called party SIP username.
Username
Calling SIP Routes calls based on the calling SIP username.
Username
Called Routes calls based on the called bearer channel properties.
Bearer
Channel
Calling Routes calls based on the calling bearer channel properties.
Bearer
Channel
Calling SIP Routes calls based on the calling SIP privacy properties.
Privacy
6. Enter the expression (related to the call properties selected in the previous step) to compare with
the call and match in order to select the route in the Expression Criteria field.
See “Routing Type” on page 338 for a list of available values for each call property.
For instance, if the property is Calling TON, you could instruct the call router to look for the following
expression:
international
If you have selected the Date/Time property in the above step, you can click the Time criteria
editor link and use the editor to easily configure the Date/Time parameters.
Figure 158: Date/Time Criteria Editor (Day Time)
• Select between the Day-Time or Time-Period settings in the Select Criteria Type drop-
down menu. If you select Time-Period, the editor changes as follows:

Figure 159: Date/Time Criteria Editor (Time Period)
• Select or enter the parameters you want, then click the Add to List button. If a
parameter is invalid (for instance, the end date is inferior to the start date), it is
displayed in red in the Time Criteria List field.
• To remove an existing parameter, select it in the Time Criteria List field, then click the
Remove Selected button.
• To update an existing parameter, select it in the Time Criteria List field, then click the
Update Selected button.
• To remove all parameters, click the Clear Parameters button.
• When done, click the Submit button.
7. If applicable, enter the name of mappings to apply to the call in the Mappings field.
You can enter more than one mapping by separating them with commas. These mappings are
executed in sequential order.
You can use the Suggestion column’s drop-down menu to select an existing mapping, if any.
The manipulations are executed before sending the call to the new destination. See “Mappings” on
If you leave this field empty, no mapping is required.
8. Select the call signalling property of the route used to modify the behaviour of the call at the SIP
protocol level in the Call Signaling drop-down menu.
You must set call signaling properties as defined in “Signalling Properties” on page 368. You can
use the Suggestion column’s drop-down menu to select between existing properties, if any.
9. Select the destination of the call when it matches in the Destination field.
The destination can be:
• route-name: The call destination is set to the route name.
• hunt-name: The call destination is set to the hunt name.
• sip-name: The call destination is set to the SIP interface name.
• isdn-name: The call destination is set to the ISDN interface name.
• r2-name: The call destination is set to the R2 interface name.
• e&m-name: The call destination is set to the E&M interface name.
• fxs-name: The call destination is set to the FXS interface name.
• fxo-name: The call destination is set to the FXO interface name.
• SipRedirect-name: When the Route source is a SIP interface, incoming SIP Invites are
replied with a 302 'Moved Temporarily' SIP response. See “SIP Redirects” on page 387
or more details.

Chapter 38 - Call Router Configuration Mappings
For instance, if you want to route calls to the hunt “CallCenter”, enter the following:
hunt-CallCenter

This brings you back to the main Call Router > Route Config web page.
You can see a yellow Yes in the Config Modified section at the top of the window. It warns you that
the configuration has been modified but not applied (i.e., the Call Router > Status differs from the
Call Router > Route Config). The Route Config sub-menu is a working area where you build up a
Call Router configuration. While you work in this area, the configured parameters are saved but not
applied (i.e., they are not used to process incoming calls). The yellow Yes flag warns you that the
configuration has been modified but is not applied.
11. Click the Apply button to enable the route.
The current routes applied are displayed in the Call Router > Status web page. You can also see
Examples
The following are some examples of routes:
Figure 160: Routes Examples
Moving a Route
Once the call router finds a routing entry that matches, it does not check the other entries, even if some of
them may still match. The routes sequence is thus very important. The call router follows the routing table rows
as they are entered in the web interface. If you want the call router to try to match one row before another one,
you must put that row first.
 To move a routing entry up or down:

2. Click the Apply button to update the Call Router > Status web page.
Deleting a Route
You can delete a routing row from the table in the web interface.
 To delete a routing entry:

Mappings
Mapping entries modify the call setup message of a call. They thus influence the routing decision and/or the
setup message leaving the call router. They are specifically called within a route.
Like the routing table, the mapping table finds the first matching entry. It then executes it by manipulating a
call property. A mapping always examines one call property and changes another property.
The call router executes all mapping entries that match by following the mapping table rows as they are
entered in the web interface. If you want the call router to try to match one row before another one, you must
put that row first.

Mappings Software Configuration Guide
The mapping may work with three types of call properties:

 calling party properties
 called party properties
 generic properties
Generic properties are used for call properties that apply to both calling and called parties.
The web interface mapping configuration is separated in two parts: Mapping Type and Mapping Expression.
You must properly configure both parts for the mapping to work as required.
When a call arrives at the mapping table, the call router proceeds as follows:
1. It examines the call property as specified in the Criteria (input) value of the Mapping Type part.
2. It selects the first matching entry.
3. It replaces the property specified in the Transformation (output) value of the Mapping Expression
part with the value of the selected entry.
Creating/Editing a Mapping Type

The Mapping Type part allows you to define the input call property to match and to define which call property
to change. The mapping type then uses one or more corresponding mapping expressions that you can define
in “Creating/Editing a Mapping Expression” on page 360.
You can add up to 40 Mapping Types.
 To create or edit a mapping type:

2. Locate the Mapping Type section.

• If you want to add a mapping type entry before an existing entry, locate the proper row
in the table and click the button of this row.
• If you want to add a mapping type entry at the end of the existing rows, click the
button at the bottom right of the Mapping Type section.
• If you want to edit an existing entry, locate the proper row in the table and click the
button.
This brings you to the Configure Mapping Type panel.

Figure 162: Configure Mapping Type Panel
4
5
6
4. Enter the name of the mapping in the Name field.

This is the name used in a route when calling a mapping. It must be unique. Aastra suggests to use
the type as part of the name for ease of identification.
There must be at least one corresponding mapping expression in the Mapping Expression table
with the exact same name. See “Creating/Editing a Mapping Expression” on page 360 for more
details.
5. Select the input call property to compare with the call and match in order to select the mapping in
the Criteria drop-down menu.
6. Select the call property to transform in the Transformation drop-down menu.
• Click the Submit button to go back to the main Call Router > Route Config web page.
You can now define a corresponding mapping expression.
• Click the Submit and Insert Expression button to directly access the proper mapping
expression dialog.
Creating/Editing a Mapping Expression

The Mapping Expression part defines the actual transformation to apply to the corresponding mapping type.
Each mapping expression must match a mapping type as defined in “Creating/Editing a Mapping Type” on
page 359.
You can add up to 100 Mapping Expressions.
 To create or edit a mapping expression:

2. Locate the Mapping Expression section.


• If you want to add a mapping expression entry before an existing entry, locate the
proper row in the table and click the button of this row.
• If you want to add a mapping expression entry at the end of the existing rows, click the
button at the bottom right of the Mapping Expression section.
button.
This brings you to the Configure Mapping Expression panel.
Figure 164: Configure Mapping Expression Panel
4
5
6
7
4. Enter the name of the mapping expression in the Name field.

This name must match a mapping type as defined in “Creating/Editing a Mapping Type” on
page 359. You can use the Suggestion column’s drop-down menu to select an existing mapping
type. When a name matches a mapping type, its type is displayed in the Type row as follows:
input type to output type
You can define several mapping expressions with the same name. In that case, the first row
matching the call is used. The rows are used in ascending order.
5. Enter the expression (related to this specific input type) to compare with the call and match in order
to select the mapping in the Criteria field.
This string differs depending on the input type selected in the Mapping Type part (Criteria drop-
down menu). For instance, if your input type is Calling TON, you could instruct the call router to look
for the following expression:
international
See “Routing Type” on page 338 for a list of available transformation values.
Table 271: Input Type Criteria
Input Type Criteria
None No criteria, always matches.

E164 If the Transformation value of the Mapping Type part is also a generic property,
this is applied to both this Calling E164 and Called E164 property.
Called E164 Selects an entry based on the called party E.164 number. You can use wildcards
to summarize entries as per “Called / Calling E164” on page 339.
Calling E164 Selects an entry based on the calling party E.164 number. You can use
wildcards to summarize entries as per “Called / Calling E164” on page 339.
Name If the Transformation value of the Mapping Type part is also a generic property,
this is applied to both this Calling Name and Called Name property.
Called Name Selects an entry based on the display name of the called party. You can use
wildcards to summarize entries as per “Called / Calling Name” on page 340.

Table 271: Input Type Criteria (Continued)
Input Type Criteria
Calling Name Selects an entry based on the display name of the calling party. You can use
wildcards to summarize entries as per “Called / Calling Name” on page 340.
TON If the Transformation value of the Mapping Type part is also a generic property,
this is applied to both this Calling TON and Called TON property.
Called TON Selects an entry based on the called party type of number as per “Called /
Calling TON” on page 339.
Calling TON Selects an entry based on the calling party type of number as per “Called /
Calling TON” on page 339.
NPI If the Transformation value of the Mapping Type part is also a generic property,
this is applied to both this Calling NPI and Called NPI property.
Called NPI Selects an entry based on the called party numbering plan indicator as per
“Called / Calling NPI” on page 339.
Calling NPI Selects an entry based on the calling party numbering plan indicator as per
“Called / Calling NPI” on page 339.
Host If the Transformation value of the Mapping Type part is also a generic property,
this is applied to both this Calling Host and Called Host property.
Called Host Selects an entry based on the remote signalling IP address or domain name of
the destination VoIP peer. You can use wildcards to summarize entries as per
“Called / Calling Host” on page 340.
Calling Host Selects an entry based on the remote signalling IP address or domain name of
the originating VoIP peer. You can use wildcards to summarize entries as per
Calling PI Selects an entry based on the presentation indicator as per “Calling PI” on
page 340.
Calling SI Selects an entry based on the screening indicator as per “Calling SI” on
page 340.
Calling ITC Selects an entry based on the information transfer capability as per “Calling ITC”
on page 341.
URI If the Transformation value of the Mapping Type part is also a generic property,
this is applied to both this Calling URI and Called URI property.
Called URI Selects an entry based on the called SIP URI properties. You can use wildcards
to summarize entries as per “Called / Calling URI” on page 340.
Calling URI Selects an entry based on the calling SIP URI properties. You can use wildcards
to summarize entries as per “Called / Calling URI” on page 340.
Date/Time Selects an entry based on the date and/or time the call arrived at the call router
as per “Date/Time” on page 341.
Phone Selects an entry based on the called or calling phone context properties as per
Context “Called / Calling Phone Context” on page 342.
Called Phone Selects an entry based on the called phone context properties as per “Called /
Context Calling Phone Context” on page 342.
Calling Selects an entry based on the calling phone context properties as per “Called /
Phone Calling Phone Context” on page 342.
Context
SIP Selects an entry based on the called or calling SIP username properties as per
Username “Called / Calling SIP Username” on page 342.

Table 271: Input Type Criteria (Continued)
Input Type Criteria
Called SIP Selects an entry based on the called SIP username properties as per “Called /
Username Calling SIP Username” on page 342.
Calling SIP Selects an entry based on the calling SIP username properties as per “Called /
Username Calling SIP Username” on page 342.
Last Selects an entry based on the last diverting reason properties as per “Last /
Diverting Original Diverting Reason” on page 342.
Reason
Last Selects an entry based on the last diverting E.164 properties as per “Last /
Diverting Original Diverting E.164” on page 343.
E164
Last Selects an entry based on the party number type of the last diverting number
Diverting properties as per “Last / Original Diverting Party Number Type” on page 343.
Party
Number Type
Last Selects an entry based on the public type of number of the last diverting number
Diverting properties as per “Last / Original Diverting Public Type Of Number” on page 343.
Public Type
Of Number
Last Selects an entry based on the private type of numbekr of the last diverting
Diverting number properties as per “Last / Original Diverting Private Type Of Number” on
Private Type page 343.
Of Number
Last Selects an entry based on the presentation of the last diverting number
Diverting properties as per “Last / Original Diverting Number Presentation” on page 344.
Number
Presentation
OriginalDiver Selects an entry based on the original diverting reason properties as per “Last /
tingReason Original Diverting Reason” on page 342.
OriginalDiver Selects an entry based on the original diverting E.164 properties as per “Last /
tingE164 Original Diverting E.164” on page 343.
Original Selects an entry based on the party number type of the original diverting number
Diverting properties as per “Last / Original Diverting Party Number Type” on page 343.
Party
Number Type
Original Selects an entry based on the public type of number of the original diverting
Diverting number properties as per “Last / Original Diverting Public Type Of Number” on
Public Type page 343.
Of Number
Called Selects an entry based on the called bearer channel properties as per “Called /
Bearer Calling SIP Username” on page 342.
Channel
Calling Selects an entry based on the calling bearer channel properties as per “Called /
Bearer Calling SIP Username” on page 342.
Channel
Calling SIP Selects an entry based on the calling SIP privacy properties as per “SIP Privacy
Privacyl Type” on page 344.
If you are editing a Date/Time property, you can click the Time criteria editor link and use the editor
to easily configure the Date/Time parameters.

Figure 165: Date/Time Criteria Editor (Day Time)
• Select between the Day-Time or Time-Period settings in the Select Criteria Type drop-
down menu. If you select Time-Period, the editor changes as follows:
Figure 166: Date/Time Criteria Editor (Time Period)
• Select or enter the parameters you want, then click the Add to List button. If a
parameter is invalid (for instance, the end date is inferior to the start date), it is
displayed in red in the Time Criteria List field.
• To remove an existing parameter, select it in the Time Criteria List field, then click the
Remove Selected button.
• To update an existing parameter, select it in the Time Criteria List field, then click the
Update Selected button.
• To remove all parameters, click the Clear Parameters button.
• When done, click the Submit button.
6. Enter the transformation (related to this specific output type) to apply in the Transformation field.
If the transformation is to replace part of an expression, it can use the matched group of the criteria.
“\0” will be replaced by the whole criteria capability and “\1” to “\9” by the matched group. See
“Groups” on page 337 for more details.
See “Routing Type” on page 338 for a list of available transformation values.
Table 272: Output Type Transformation
Output Type Transformation
None No transformation is applied.

Table 272: Output Type Transformation (Continued)
E164 If the Criteria value of the Mapping Type part is also a generic property, this is
applied to both the Calling E164 and Called E164 properties.
Called E164 Modifies the called party E.164 number as per “Called / Calling E164” on
page 339.
Calling E164 Modifies the calling party E.164 number as per “Called / Calling E164” on
page 339.
Name If the Criteria value of the Mapping Type part is also a generic property, this is
applied to both the Calling Name and Called Name properties.
Called Name Sets the display name of the called party as per “Called / Calling Name” on
page 340.
Calling Name Sets the display name of the calling party as per “Called / Calling Name” on
page 340.
TON If the Criteria value of the Mapping Type part is also a generic property, this is
applied to both the Calling TON and Called TON properties.
Called TON Sets the called party type of number as per “Called / Calling TON” on page 339.
Calling TON Sets the calling party type of number as per “Called / Calling TON” on page 339.
NPI If the Criteria value of the Mapping Type part is also a generic property, this is
applied to both the Calling NPI and Called NPI properties.
Called NPI Sets the called party numbering plan indicator as per “Called / Calling NPI” on
page 339.
Calling NPI Sets the calling party numbering plan indicator as per “Called / Calling NPI” on
page 339.
Host If the Criteria value of the Mapping Type part is also a generic property, this is
applied to both the Calling Host and Called Host properties.
Called Host Sets the remote IP address or domain name of the destination VoIP peer as per
Calling Host Sets the remote IP address or domain name of the originating VoIP peer as per
Calling PI Sets the presentation indicator as per “Calling PI” on page 340.
Calling SI Sets the screening indicator as per “Calling SI” on page 340.
Calling ITC Sets the information transfer capability as per “Calling ITC” on page 341.
URI If the Criteria value of the Mapping Type part is also a generic property, this is
applied to both the Calling URI and Called URI properties.
Called URI Sets the called URI as per “Called / Calling URI” on page 340.
Calling URI Sets the calling URI as per “Called / Calling URI” on page 340.
Phone If the Criteria value of the Mapping Type part is also a generic property, this is
Context applied to both the Calling Phone Context and Called Phone Context properties.
Called Phone Sets the called Phone Context as per “Called / Calling Phone Context” on
Context page 342.
Calling Sets the calling Phone Context as per “Called / Calling Phone Context” on
Phone page 342.
Context

SIP If the Criteria value of the Mapping Type part is also a generic property, this is
Username applied to both the Calling SIP Username and Called SIP Username properties.
Called SIP Sets the called SIP Username as per “Called / Calling SIP Username” on
Username page 342.
Calling SIP Sets the calling SIP Username as per “Called / Calling SIP Username” on
Username page 342.
Last Sets the last diverting reason properties as per “Last / Original Diverting
Diverting Reason” on page 342.
Reason
Last Sets the last diverting E.164 properties as per “Last / Original Diverting E.164”
Diverting on page 343.
E164
Last Sets the party number type of the last diverting number properties as per “Last /
Diverting Original Diverting Party Number Type” on page 343.
Party
Number Type
Last Sets the public type of number of the last diverting number properties as per
Diverting “Last / Original Diverting Public Type Of Number” on page 343.
Public Type
Of Number
Last Sets the private type of number of the last diverting number properties as per
Diverting “Last / Original Diverting Private Type Of Number” on page 343.
Private Type
Of Number
Last Sets the presentation of the last diverting number properties as per “Last /
Diverting Original Diverting Number Presentation” on page 344.
Number
Presentation
Original Sets the original diverting reason properties as per “Last / Original Diverting
Diverting Reason” on page 342.
Reason
Original Sets the original diverting E.164 properties as per “Last / Original Diverting
Diverting E.164” on page 343.
E164
Original Sets the party number type of the original diverting number properties as per
Diverting “Last / Original Diverting Party Number Type” on page 343.
Party
Number Type
Original Sets the public type of number of the original diverting number properties as per
Diverting “Last / Original Diverting Public Type Of Number” on page 343.
Public Type
Of Number
Original Sets the private type of number of the original diverting number properties as per
Diverting “Last / Original Diverting Private Type Of Number” on page 343.
Private Type
Of Number

Original Sets the Presentation of the original diverting number properties as per “Last /
Diverting Original Diverting Number Presentation” on page 344.
Number
Presentation
Called Sets the called bearer channel properties as per “Called / Calling SIP
Bearer Username” on page 342.
Channel
Calling Sets the calling bearer channel properties as per “Called / Calling SIP
Bearer Username” on page 342.
Channel
Debug Reserved for debug configuration.
You cannot use Date/Time as an output type transformation.

7. If applicable, enter the name of one or more subsequent mappings to execute in the Sub Mappings
field.
You can enter more than one mapping by separating them with commas. The mappings are
executed in sequential order.
You can use the Suggestion column’s drop-down menu to select between existing values, if any.
You may want to send the result of the first mapping to another one. Once the subsequent mapping
is finished, the call router continues to check the mapping entries for matching entries. For instance,
if the call router is checking the fourth mapping entry and that entry uses subsequent mapping, the
call router executes the subsequent mapping, then resumes checking the fifth mapping entry, and
so on.
The maximal number of subsequent interleaved mapping is 3.
• Click the Submit button to go back to the main Call Router > Route Config web page.
You can see a yellow Yes in the Config Modified section at the top of the window. It
warns you that the configuration has been modified but not applied (i.e., the Call Router
> Status differs from the Call Router > Route Config). The Route Config sub-menu is a
working area where you build up a Call Router configuration. While you work in this area,
the configured parameters are saved but not applied (i.e., they are not used to process
incoming calls). The yellow Yes flag warns you that the configuration has been modified
but is not applied.
• Click the Submit and Insert Expression button to create another expression for the
same type.
9. Click the Apply button to enable the mapping entry.
The current mappings applied are displayed in the Call Router > Status web page. You can also
Examples
The following are some examples of mappings:

Chapter 38 - Call Router Configuration Signalling Properties
Figure 167: Mappings Examples
Moving a Mapping Type or Expression Row

The mapping entries sequence is very important. The call router follows the mapping table rows as they are
entered in the web interface. If you want the call router to try to match one row before another one, you must
put that row first.
 To move a mapping entry up or down:

1. In the Mapping Type or Mapping Expression table, either click the or arrow of the row you
want to move until the entry is properly located.
Deleting a Mapping Type or Expression Row

You can delete a mapping row from the Mapping Type or Mapping Expression table in the web interface.
 To delete a mapping entry:

Signalling Properties
Standards Supported • RFC 3323: A Privacy Mechanism for the Session Initiation
Protocol (SIP) (only supports 'none' as Privacy level)
• RFC 3325: Private Extensions to the Session Initiation
Protocol (SIP) for Asserted Identity within Trusted Networks
(supports 'id' as Privacy level. Accept/send P-Asserted-
Identity and P-Preferred-Identity.)
Call signalling specifies how to set up a call to the destination Aastra unit or 3rd party equipment. Call signalling
properties are assigned to a route and used to modify the behaviour of the call at the SIP protocol level.
Signaling Properties are applied after mappings rules.
Like the routing table, the signalling properties table finds the first matching entry. It then executes it by
modifying the behaviour of the call.
You can add up to 40 Signalling Properties.
Creating/Editing a Signalling Property

The web interface allows you to create a signalling property or modify the parameters of an existing one. The
signalling properties are called from a route as described in “Routes” on page 353.

Signalling Properties Software Configuration Guide
 To create or edit a signalling property:

2. Locate the Signaling Properties section.

• If you want to add a signalling property entry before an existing entry, locate the proper
• If you want to add a signalling property entry at the end of existing rows, click the
button at the bottom right of the Signaling Properties section.
button.
This brings you to the Configure Signaling Properties panel.
Figure 169: Configure Signaling Properties Panel
4
5
6
7
8
9
10
11
12
4. Enter the name of the signalling property in the Name field.

The name must be unique. It will be used in routes to call a specific signalling property as described
in “Routes” on page 353.
5. Select whether or not the early connect feature is enabled in the Early Connect drop-down menu.
When early connect is enabled, the SIP call is connected by sending a 200 OK message instead of
a 183 Session Progress message with early media, if the called party answers the call. It allows
interoperability with units that do not support the 183 Session Progress with SDP message.
When early connect is disabled, call progress tones or announcements are transmitted in the early
SIP dialog.
6. Select whether or not the early disconnect feature is enabled in the Early Disconnect drop-down
menu.
This feature is useful to avoid hearing the end of call tone when the far end party terminates the call
during a conference.

Chapter 38 - Call Router Configuration Signalling Properties
When early disconnect is:

• enabled, the SIP BYE message is sent upon receiving the ISDN “Disconnect” signal.
• disabled, the SIP BYE message is sent upon receiving the ISDN “Call release” signal.
If early disconnect is enabled but no ISDN “Disconnect” message is received, the SIP BYE
message is sent upon receiving an ISDN “Call release” signal as if the early disconnect was
disabled.
7. Define the SIP messages destination (where an INVITE is sent) in the Destination Host field.
It can override the Called Host property set by a mapping rule because signalling properties are
applied after mappings.
You can also use the macro local_ip_port to replace the properties by the local IP address and
port of the listening network of the SIP gateway used to send the INVITE.
8. Define whether or not to enable the 180 with SDP allowed feature in the Allow 180 SDP drop-down
menu.
Table 273: 180 with SDP Parameters
Enable The unit can send a SDP in the provisional response 180. Thus when the ISDN
peer sends an alerting with indication to open the voice (or if the voice is already
opened), the unit sends a 180 with SDP. This is the default value.
Disable A SIP 183 with SDP is sent instead of a 180 with SDP. This does not affect the 180
without SDP. This is useful if your proxy has issues receiving 180 with SDP
messages.
The SIP 183 with SDP replacing the SIP 180 with SDP is not sent if a 183 with
SDP has already been sent.
9. Define whether or not to enable the 183 without SDP allowed feature in the Allow 183 No SDP drop-
down menu.
Table 274: 183 without SDP Parameters
Enable When enabled, the unit sends a 183 without SDP upon receiving an ISDN
progress indicator without any indication to open a voice stream. This is the default
value.
Disable When disabled, nothing is sent instead of a 183 without SDP. This does not affect
the 183 with SDP. This is useful if your proxy has issues receiving 183 without
SDP messages.
10. Set the privacy level of the call in the Privacy drop-down menu.
Table 275: Privacy Levels
Level Description Effects on incoming SIP call Effects on outgoing SIP call
Disable No privacy None None

is used.

Signalling Properties Software Configuration Guide
Table 275: Privacy Levels (Continued)
Level Description Effects on incoming SIP call Effects on outgoing SIP call
None Use P- None Adds two headers:

Asserted • Privacy: none
Identity
• P-Asserted-Identity:
privacy.
p_asserted_identity_v
alue
p_asserted_identity_value is the
call's From URI unless a SIP
header translation has been added
to the Signaling Properties for the
Identity-header.
Id Use P- The calling-name is empty and the Always adds one header:
Preferred PI is set to restricted. • P-Preferred-Identity:
Identity p_preferred_identity_v
privacy. alue
p_ preferred _identity_value is the
call's From URI unless a SIP
header translation has been added
to the Signaling Properties for the
Identity-header.
If the incoming call's PI property is
restricted, another header is
added:
• Privacy: id
Rpid Use None One header always added :
Remote- • Remote-Party-ID:
Party-ID remote_party_id_value
privacy.
"Optional Friendly
Name"<sip:[email protected]>
;party=calling
Where remote_party_id_value
should be set by the SIP Headers
Translation. It consists of an
optional friendly name followed by
the SIP URI and the party
direction.
Example:
Remote-Party-ID: "John
Doe"<sip:[email protected].
12>;party=calling
11. Enter the name of one or more SIP headers translation to apply to the call in the SIP Headers
Translations field.
You must define SIP headers translations as defined in “SIP Headers Translations” on page 372.
You can use the Suggestion column’s drop-down menu to select between existing translations, if
any.
You can enter more than one translation. In that case, the translations are separated with “,” and
are executed in sequential order.
12. Enter the name of one or more call properties translation to apply to the call in the Call Properties
Translations field.

Chapter 38 - Call Router Configuration SIP Headers Translations
You must set call properties translations as defined in “Call Properties Translations” on page 376.
You can use the Suggestion column’s drop-down menu to select between existing translations, if
any.
You can enter more than one translation. In that case, the translations are separated with “,” and
are executed in sequential order.
14. Click the Apply button to enable the signalling property entry.
The current properties applied are displayed in the Call Router > Status web page. You can also
Examples
The following are some examples of signalling properties:
Figure 170: Signalling Properties Examples
Moving a Signalling Property Row

The signalling properties entries sequence is very important. The call router follows the signalling properties
table rows as they are entered in the web interface. If you want the call router to try to match one row before
another one, you must put that row first.
 To move a signalling property entry up or down:

Deleting a Signalling Property Row

You can delete a signalling property row from the table in the web interface.
 To delete a signalling property entry:

SIP Headers Translations
A SIP Headers Translation overrides the default value of SIP headers in an outgoing SIP message. It modifies
the SIP headers before the call is sent to its destination.

SIP Headers Translations Software Configuration Guide
Like the routing table, the SIP headers translation table finds the first matching entry. It then executes it by
You can add up to 100 SIP Headers Translations.
Creating/Editing a SIP Headers Translation

The web interface allows you to create a SIP header translation or modify the parameters of an existing one.
The SIP headers translations are called from a signalling property as described in “Signalling Properties” on
page 368.
 To create or edit a SIP headers translation:

2. Locate the SIP Headers Translations section.

• If you want to add a SIP headers translation before an existing entry, locate the proper
• If you want to add a SIP headers translation at the end of existing rows, click the
button at the bottom right of the SIP Headers Translations section.
button.
This brings you to the Configure SIP Headers Translation panel.
Figure 172: Configure SIP Headers Translation Panel
4
5
6
7
4. Enter the name of the SIP headers translation in the Name field.
5. Set which SIP header is modified by this translation in the SIP Header drop-down menu.
Table 276: SIP Headers
SIP Header Description
From Header (Host Part) Host part of the From header's URI.

Chapter 38 - Call Router Configuration SIP Headers Translations
Table 276: SIP Headers (Continued)
SIP Header Description
From Header (User Part) User part of the From header's URI.
Identity Header (Host Part) Host part of the Identity header's URI.
Identity Header (User Part) User part of the Identity header's URI.
Identity Header (Phone Number) Phone number in the Identity header's tel URL.
Request Line (Host Part) Host part of the Request line's URI.
Request Line (User Part) User part of the Request line's URI.
To Header (Host Part) Host part of the To header's URI.
To Header (User Part) User part of the To header's URI.
6. Set what information is used to build the selected SIP header in the Built From drop-down menu.
Table 277: Built From Information
Built From Description
Called E164 Use the called party E.164 property.

Destination Host Use the destination host configured in the signalling
properties of which this translation is part.
Domain Use the domain name configured in the unit.
Fix Value Use a fix value as defined in the Fix Value field (see Step 7).
Host Name Use the host name configured in the unit.
Local Ip Use the local IP address.
Calling Bearer Channel Use the calling bearer channel.
SIP Endpoint Username Use the SIP username associated with the endpoint.
Calling Name Use the calling party name property.
Calling E164 Use the calling party E.164 property.
7. If you have selected Fix Value in the Built From drop-down menu, enter a fix value to be inserted
in the SIP header in the Fix Value field.
For instance, you could hide the caller’s name in a SIP message by using the From Header (User
Part) SIP header and entering “anonymous” in the Fix Value field.
9. Click the Apply button to enable the SIP headers translation entry.
Example

SIP Headers Translations Software Configuration Guide
The following is an example of SIP headers translations:
Figure 173: SIP Headers Translations Example
Moving a SIP Headers Translation Row

The SIP headers translation entries sequence is very important. The signalling properties table follows the SIP
headers translation table rows as they are entered in the web interface. If you want the signalling properties
table to try to match one row before another one, you must put that row first.
 To move a SIP headers translation entry up or down:

Deleting a SIP Headers Translation Row

You can delete a SIP headers translation row from the table in the web interface.
 To delete a SIP headers translation entry:


Chapter 38 - Call Router Configuration Call Properties Translations
Call Properties Translations
A Call Properties Translation overrides the default value of call properties in an incoming SIP message. It
modifies the call properties before the call is sent to its destination.
Like the routing table, the call properties translation table finds the first matching entry. It then executes it by
You can add up to 100 Call Properties Translations.
Creating/Editing a Call Properties Translation

The web interface allows you to create a call properties translation or modify the parameters of an existing
one. The call properties translations are called from a signalling property as described in “Signalling
Properties” on page 368.
 To create or edit a call properties translation:

2. Locate the Call Properties Translations section.

• If you want to add a call properties translation before an existing entry, locate the
proper row in the table and click the button of this row.
• If you want to add a call properties translation at the end of existing rows, click the
button at the bottom right of the Call Properties Translations section.
button.
This brings you to the Configure Call Properties Translation panel.
Figure 175: Configure Call Properties Translation Panel
4
5
6
7
4. Enter the name of the call properties translation in the Name field.

Call Properties Translations Software Configuration Guide
5. Set which call property is modified by this translation in the Call Property drop-down menu.
Table 278: Call Properties
Call Property Description
Called E164 Called party E.164 property.

Calling E164 Calling party E.164 property.
Called Name Called party name property.
Calling Name Calling party name property.
Called Uri Called URI name property.
Calling Uri Calling URI name property.
Called Bearer Channel Called bearer channel property.
6. Set what information is used to build the selected call property in the Built From drop-down menu.
Table 279: Built From Information
Built From Description
Domain Use the domain name configured in the unit.

Fix Value Use a fix value as defined in the Fix Value field (see Step 7).
From Header (Uri) Use the From header's URI.
From Header (Friendly Name) Use the friendly name part of the From header.
From Header (User Part) Use the user part of the From header's URI.
Identity Header (Uri) Use the Identity header's URI.
Identity Header (User Part) Use the user part of the Identity header's URI.
Identity Header (Phone Number) Use the phone number in the Identity header's tel URL. The
phone number is not retrieved if the received tel URL is
invalid. Only the phone number part is retrieved. Examples:
• Received header: P-Preferred-Identity:
<tel:8298749;phone-context=819>
Retrieved phone number: 8298749
<tel:+8298749>
Retrieved phone number: 8298749
<tel:8298749>
Retrieved phone number: None, the received header
is invalid.
Identity Header (Friendly Name) Use the friendly name in the Identity header's URI.
Local Ip Use the local IP address.
Request Line (Uri) Use the Request line's URI.
Request Line (User Part) Use the user part of the Request line's URI.
To Header (Uri) Use the To header's URI.
To Header (Friendly Name) Use the friendly name part of the To header.
To Header (User Part) Use the user part of the To header's URI.

Chapter 38 - Call Router Configuration Call Properties Translations
7. If you have selected Fix Value in the Built From drop-down menu, enter a fix value to be inserted
in the call property in the Fix Value field.
For instance, you could hide the callee’s name in a SIP message by using the From Header (User
Part) SIP header and entering “anonymous” in the Fix Value field.
9. Click the Apply button to enable the call properties translation entry.
Example
The following is an example of call properties translations:
Figure 176: Call Properties Translations Example
Moving a Call Properties Translation Row

The call properties translation entries sequence is very important. The signalling properties table follows the
call properties translation table rows as they are entered in the web interface. If you want the signalling
properties table to try to match one row before another one, you must put that row first.
 To move a call properties translation entry up or down:

Deleting a Call Properties Translation Row

You can delete a call properties translation row from the table in the web interface.
 To delete a SIP headers translation entry:


Hunt Service Software Configuration Guide
Hunt Service
Routes and mappings only manipulate address properties of a call. The hunt service hunts an incoming call
to multiple interfaces. It accepts a call routed to it by a route or directly from an interface and creates another
call that is offered to one of the configured destination interfaces. If this destination cannot be reached, the
hunt tries another destination until one of the configured destinations accepts the call. When an interface
accepts a call, the interface hunting is complete and the hunt service merges the original call with the new call
to the interface that accepted the call.
The hunt sequence is very important. The call router follows the hunt rows as they are entered in the web
interface. If you want the call router to try to match one row before another one, you must put that row first.
You can add up to 40 Hunts.
Creating/Editing a Hunt
The web interface allows you to create a hunt or modify the parameters of an existing one.
 To create or edit a hunt:

2. Locate the Hunt section.

• If you want to add a hunt entry before an existing entry, locate the proper row in the
table and click the button of this row.
• If you want to add a hunt entry at the end of existing rows, click the button at the
bottom right of the Hunt section.
button.
This brings you to the Configure Hunt panel.

Chapter 38 - Call Router Configuration Hunt Service
Figure 178: Configure Hunt Panel
6
7
8
4. Enter the name of the hunt in the Name field.

The name must be unique. If more than one hunt have the same name, only the first hunt is used.
5. Define a list of hunt destinations separated by commas in the Destinations field.
This is the interface, route, or hunt that is tried during the hunt’s interface hunting. The destination
can either be:
• route-name: The call destination is the route name.
• hunt-name: The call destination is the hunt name.
• sip-name: The call destination is the SIP interface name.
• isdn-name: The call destination is the ISDN interface name.
• r2-name: The call destination is the R2 interface name.
• e&m-name: The call destination is the E&M interface name.
• fxs-name: The call destination is the FXS interface name.
• fxo-name: The call destination is the FXO interface name.
Only FXS interfaces are supported if the selection algorithm Simultaneous is used (see Step 6).
6. Select the algorithm used to select the order of the destination in the Selection Algorithm drop-down
menu.
The algorithm can be:
• Sequential: The hunt tries the destination in the same order as listed. The first
destination hunted is the first listed.
• Cyclic: The Aastra unit starts from the destination that follows the destination used for
the last hunt. Subsequent calls try another first destination in a round-robin method. For
instance, if the destination is set to 'x, y, z', the destination the hunt tries is in the
following order:
1. x,y,z
2. y,z,x
3. z,x,y
4. x,y,z
• Simultaneous: The hunt tries every available destination at the same time. The first
destination to pick up has the call. Other destinations stop ringing. This method can
only have FXS endpoints as destinations.
7. Set the maximal time, in seconds (s), allowed to an interface to handle the call in the Timeout field.

After this timeout has elapsed, the next destination is tried when the current destination does not
answer. This feature is useful to ensure a minimal time of response and fallback to other
destinations. Some interfaces (e.g. SIP, which has a default timeout of 32 seconds) may wait an
arbitrary long time until an answer is returned.
Note: This parameter is not applicable if the selection algorithm Simultaneous is used (see Step 6).
Setting the field to 0 disables the timeout, which means that the call router waits indefinitely for the
interface to respond. This does not affect the internal interface timeouts (the ISDN timeout as
defined in ITU norms or the SIP transmission timeout) that will eventually stop the call and the call
router will try another destination.
Example:
You want a call from ISDN to SIP to fallback to another ISDN interface when the SIP destination
cannot be contacted within 5 seconds.
You thus create a hunt with the following destinations in order:
sip-[gateway name], isdn-[fallback interface]
and set the timeout to 5. The Selection Algorithm drop-down menu must be set to Sequential to
always try the SIP destination first.
Figure 179: Hunt Timeout Example
The Aastra unit has the following behaviour if the SIP transmission timeout has the default value
(32 seconds):
a. A new call comes from an ISDN interface and the call router sets the destination of the call to
the isdn-to-sip hunt.
b. The call router starts the hunt timeout (5 s) and tries the first destination sip-default.
c. The SIP interface performs a DNS query to resolve the server name. The DNS result returns
server A and server B.
d. The SIP interface sends an INVITE to the server A.
e. The hunt timeout elapses, so the call router cancels the call to the SIP interface and tries the
second destination isdn-Slot3/Bri2. The hunt timeout is restarted.
f. The SIP interface continues to send the INVITE retransmission until the SIP transmission
timeout elapses. RFC 3261 states that an INVITE request cannot be cancelled until the
destination sends a response. If the destination responds before the SIP transmission timeout
elapses, a CANCEL or BYE request is sent. The SIP interface will not try to use the server B
location.
The Aastra unit has the following behaviour if the SIP transmission timeout is set to 3 seconds:
a. A new call comes from an ISDN interface and the call router sets the destination of the call to
the isdn-to-sip hunt.
b. The call router starts the hunt timeout (5 s) and tries the first destination sip-default.
c. The SIP interface performs a DNS query to resolve the server name. The DNS result returns
server A and server B.
d. The SIP interface sends an INVITE to the server A.
e. A SIP transmission timeout occurs after 4 seconds and the SIP interface sends an INVITE to
the server B.
f. The hunt timeout elapses, so the call router cancels the call to the SIP interface and tries the
second destination isdn-Slot3/Bri2. The hunt timeout is restarted.
g. The SIP interface continues to send the INVITE retransmission until the SIP transmission
timeout elapses. RFC 3261 states that an INVITE request cannot be cancelled until the

destination sends a response. If the destination responds before the SIP transmission timeout
elapses, a CANCEL or BYE request is sent.
Note: The maximal response time of a SIP interface is the transmission timeout total of all SIP destination
locations + the DNS query time.
The SIP transmission timeout can be set in the Transmission Timeout field of the SIP Interop
section, SIP > Interop page (“SIP Interop” on page 312).
8. Select call rejection causes to continue the hunt in the Causes field.
When an interface has a problem placing a call to the final destination, it drops the call by specifying
a drop cause based on Q.850 ISUP drop causes. Separate the causes with commas.
See “Call Rejection (Drop) Causes” on page 382 for a list of drop causes.
Note: This parameter is not applicable if the selection algorithm Simultaneous is used (see Step 6).

10. In the main Call Routing Config web page, click the Apply button to enable the hunt.
The current hunts applied are displayed in the Call Router > Status web page. You can also see
Examples
The following are some examples of hunts:
Figure 180: Hunt Example
Call Rejection (Drop) Causes

When a destination interface drops the call, the hunt service must supply a call rejection cause based on Q.850
ISUP drop causes. The Aastra unit offers the following drop causes categories:
 Normal Event
 Resource Unavailable
 Service or Option Not Available
 Service or Option Not Implemented
 Invalid Message
 Protocol Error
 Interworking
Note: You can use any custom code between 1 and 127.

Normal Event
The following table lists all normal events drop causes. These causes are used to drop the original call.
Table 280: Normal Event Drop Causes
# Cause Description
1 Unassigned The calling user requested a destination that cannot be reached because
(unallocated) number the number is unassigned.
2 No route to specified The destination is asked to route the call through an unrecognized network.
transit network This may mean that:
• The wrong transit network code was dialed.
• The transit network does not serve this equipment.
• The transit network does not exist.
3 No route to destination The called party cannot be reached because the network through which the
call has been routed does not serve the destination address.
6 Channel unacceptable The sending entity cannot accept the channel most recently identified for
use in this call.
7 Call awarded and The user has been awarded the incoming call, which is being connected to a
being delivered in an channel already established to that user for similar calls.
established channel
16 Normal call clearing The call is being cleared because one of the users involved with the call has
requested that the call be cleared (usually, a call participant hung up).
17 User busy The called party is unable to accept another call because all channels are in
use. It is noted that the user equipment is compatible with the call.
18 No user responding The called party does not respond to a call establishment message with
either an alerting or connect indication within the time allotted. The number
that is being dialed has an active D-channel, but the far end chooses not to
answer.
19 User alerting, no The called party has been alerted but does not respond with a connect
answer indication within the time allotted.
21 Call rejected The remote equipment can accept the call but rejects it for an unknown
reason, although it could have accepted it because the equipment sending
this cause is neither busy nor incompatible.
22 Number changed The called number indicated by the calling party is no longer assigned.
26 Non-selected user The user has not been awarded the incoming call.
clearing
27 Destination out of The destination indicated by the user cannot be reached because the
order destination's interface is not functioning correctly. This can be a temporary
condition, but it could last for an extended period.
28 Invalid number format The called party cannot be reached because the called party number is not
(incomplete number) in a valid format or is not complete.
29 Facility rejected The network cannot provide the facility requested by the user.
30 Response to STATUS The STATUS message is generated in direct response to receiving a
ENQUIRY STATUS ENQUIRY message.
31 Normal, unspecified Reports a normal event only when no other cause in the normal class
applies.
Resource Unavailable

The following table lists all resource unavailable drop causes. These causes are used to hunt the next
destination.
Table 281: Resource Unavailable Drop Causes
# Cause Description
34 No circuit/channel There is no appropriate circuit or channel presently available to handle the

available call (usually, no B-channels are available to make the selected call).
38 Network out of order The network is not functioning properly and the condition is likely to last for
an extended period.
41 Temporary failure The network is not functioning properly and the condition should be resolved
quickly.
42 Switching equipment Cannot reach the destination because the network switching equipment is
congestion temporary experiencing high traffic.
43 Access information The network could not deliver access information to the remote user as
discarded requested.
44 Requested circuit/ The other side of the interface cannot provide the circuit or channel
channel not available indicated by the requested entity.
47 Resource unavailable, The requested channel or service is unavailable for an unknown reason.
unspecified
Service or Option Not Available

The following table lists all service or option not available drop causes. These causes are used to drop the
original call.
Table 282: Service or Option Not Available Drop Causes
# Cause Description
57 Bearer capability not The user has requested a bearer capability that is implemented on the
authorized equipment but the user is not authorized to use it.
58 Bearer capability not The user has requested a bearer capability that is implemented by the
presently available equipment and is currently unavailable.
63 Service or option not The network or remote equipment cannot provide the requested service
available, unspecified option for an unspecified reason.
Service or Option Not Implemented

The following table lists all service or option not implemented drop causes. These causes are used to drop the
original call.
Table 283: Service or Option Not Implemented Drop Causes
# Cause Description
65 Bearer capability not The remote equipment does not support the requested bearer capability.
implemented
66 Channel type not The remote equipment does not support the requested channel type.
implemented
69 Requested facility not The remote equipment does not support the requested supplementary
implemented service.

Table 283: Service or Option Not Implemented Drop Causes (Continued)
# Cause Description
70 Only restricted digital The calling party has requested an unrestricted bearer service but the
information bearer remote equipment only supports the restricted version of the requested
capability is available bearer capacity.
79 Service or option not The network or remote equipment cannot provide the requested service
implemented, option for an unspecified reason. This can be a subscription problem.
unspecified
Invalid Message
The following table lists all invalid message drop causes. These causes are used to drop the original call.
Table 284: Invalid Message Drop Causes
# Cause Description
81 Invalid call reference The remote equipment has received a message with a call reference that is
value not currently in use on the user-network interface.
82 Identified channel Indicates a call attempt on a channel that is not configured.
does not exist
83 A suspended call Attempted to resume a call with a call identity that differs from the one in use
exists, but this call for any presently suspended calls.
identity does not
84 Call identity in use The network has received a call suspended request containing a call identity
that is already in use for a suspended call.
85 No call suspended The network has received a call resume request containing a call identity
information element that does not indicate any suspended call.
86 Call having the The network has received a call identity information element indicating a
requested call identity suspended call that has in the meantime been cleared while suspended.
has been cleared
88 Incompatible The remote equipment has received a request to establish a call with
destination compatibility attributes that cannot be accommodated.
91 Invalid transit network Received a transit network identification of an incorrect format was received.
selection
95 Invalid message, Received an invalid message event.
unspecified
Protocol Error
The following table lists all protocol error drop causes. These causes are used to drop the original call.
Table 285: Protocol Error Drop Causes
# Cause Description
96 Mandatory information The remote equipment has received a message that is missing an
element is missing information element (IE). This IE must be present in the message before the
message can be processed.
97 Message type non- The remote equipment has received a message with a missing information
existent or not element that must be present in the message before the message can be
implemented processed.

Table 285: Protocol Error Drop Causes (Continued)
# Cause Description
98 Message not The remote equipment has received a message that is not allowed while in
compatible with call the current call state.
state or message type
non-existent or not
implemented
99 Information element The remote equipment has received a message that includes information
non-existent or not elements or parameters that are not recognized.
implemented
100 Invalid information The remote equipment has received a message that includes invalid
element contents information in the information element or call property.
101 Message not Received an unexpected message that is incompatible with the call state.
compatible with call
state
102 Recovery on time A procedure has been initiated by the expiration of a timer in association
expiry with error handling procedures.
111 Protocol error, An unspecified protocol error with no other standard cause occurred.
unspecified
Interworking
The following table lists all interworking drop causes. These causes are used to drop the original call.
Table 286: Interworking Drop Causes
# Cause Description
127 Interworking, An event occurs, but the network does not provide causes for the action it
unspecified takes. The precise problem is unknown.
Moving a Hunt
The hunt sequence is very important. The call router follows the hunt rows as they are entered in the web
interface. If you want the call router to try to match one row before another one, you must put that row first.
 To move a hunt entry up or down:

Deleting a Hunt
You can delete a hunt row from the table in the web interface.
 To delete a hunt entry:

1. Click the button of the row you want to move.

SIP Redirects Software Configuration Guide
SIP Redirects
The SIP Redirect allows SIP redirections to be configured. These SIP Redirect entries can be used as
destinations in route rules. This type of destination is valid only when the Source of the route rule is a SIP
interface.
When a route rule is configured with a SIP Redirect destination, incoming SIP Invites are replied with a 302
"Moved Temporarily" SIP response.
Creating/Editing a SIP Redirect

The web interface allows you to create a SIP Redirect or modify the parameters of an existing one.
 To create or edit a SIP Redirect:

2. Locate the SIP Redirects section.

• If you want to add a SIP Redirect entry before an existing entry, locate the proper row in
the table and click the button of this row.
• If you want to add a SIP Redirect entry at the end of existing rows, click the button
at the bottom right of the SIP Redirects section.
button.
This brings you to the Configure SIP Redirect panel.
Figure 182: Configure SIP Redirect Panel
4
5
4. Enter the name of the SIP Redirect in the Name field.

The name must be unique. If more than one SIP Redirect have the same name, only the first SIP
Redirect is used.
5. Set the Destination Host field with the host address inserted in the Moved Temporarily response.

Chapter 38 - Call Router Configuration Configuration Examples
7. In the main Call Routing Config web page, click the Apply button to enable the SIP Redirect.
The current SIP Redirects applied are displayed in the Call Router > Status web page. You can also
Examples
The following are some examples of SIP Redirects:
Figure 183: SIP Redirects Example
Moving a SIP Redirect

The SIP Redirect sequence is very important. The call router follows the SIP Redirect rows as they are entered
in the web interface. If you want the call router to try to match one row before another one, you must put that
row first.
 To move a SIP Redirect entry up or down:

Deleting a SIP Redirect

You can delete a SIP Redirect row from the table in the web interface.
 To delete a SIP Redirect entry:

1. Click the button of the row you want to move.
Configuration Examples
The following are examples of configuration you could do with the call router.

Configuration Examples Software Configuration Guide
Figure 184: Configuration Examples

Chapter 38 - Call Router Configuration Configuration Examples

C H A P T E R
39 Auto-Routing Configuration
This chapter describes the auto-routing feature.
Auto-Routing
The auto-routing feature is an aid to call routing configuration. When this feature is enabled, routing rules are
automatically generated for all endpoints marked as "Auto-routable". For each auto-routable endpoint, two
rules are generated and added to the Call Router: one directing incoming calls from the associated auto-
routing SIP gateway to the endpoint, and one sending outgoing calls from the endpoint to the associated auto-
routing SIP gateway.
The auto-routing routes are not displayed in the Route Configuration page because you cannot edit them.
They are however listed in the Status page and are attributed a type:
 User: the route has been manually entered by the user.
 Auto: this is an auto-routing route.
Note: Auto-routing can only be used if the username of the endpoint is an E.164 string and the username
part of the request-URI of the received INVITE can be converted into an E.164. See “Manual Routing” on
 To activate auto-routing:
1. In the web interface, click the Call Router link, then the Auto-routing sub-link.
Figure 185: Call Router – Auto-Routing Web Page
2
3
4
5
6
7
2. In the top section, set the Auto-routing drop-down menu with the proper behaviour.
If you select Enable, routes are automatically added to the Route Table in order to connect the
endpoints marked as eligible for auto-routing (see Step 3) and the designated SIP gateway (see
Step 4). These automatic routes are displayed in the Call Router > Status page, but do not show up
in the Call Router > Route Configuration page.

Chapter 39 - Auto-Routing Configuration Auto-Routing
3. Select the type of criteria to use to create automatic rules from SIP to the telephony endpoints in the
Criteria Type drop-down menu.
Table 287: Criteria Types
E164 The E.164 associated with the endpoint is used as criterion.

Sip Username The SIP username associated with the endpoint is used as
criterion.
4. Set the Incoming Mappings field with the name of the properties manipulations associated with the
route from the SIP gateway to the endpoint.
You can specify more than one mapping by separating them with ','. They are executed in sequential
order. See “Mappings” on page 484 for more details.
5. Set the Outgoing Mappings field with the name of the properties manipulations associated with the
route from the endpoint to the SIP gateway.
You can specify more than one mapping by separating them with ','. They are executed in sequential
order. See “Mappings” on page 484 for more details.
6. Set the Incoming Signaling Properties field with the name of the signaling properties associated with
the route from the SIP gateway to the endpoint.
See “Signalling Properties” on page 494 for more details.
7. Set the Outgoing Signaling Properties field with the name of the signaling properties associated with
the route from the endpoint to the SIP gateway.
See “Signalling Properties” on page 494 for more details.
8. Click the Submit button to enable auto-routing.
The current routes applied are displayed in the Call Router > Status web page. They are added at
the end of the routes that are already present, if any. This ensures that the user-defined routes
always have precedence over the automatic routes when both types of routes apply to the same
endpoint.
Endpoints Auto-Routing
This section allows you to link an endpoint to several SIP gateways.
 To set Endpoints auto-routing parameters:

1. In the Endpoints auto-routing section of the Auto-routing page, locate the proper endpoint in the
table and click the button.
The Configure Auto-Routing page displays:

Auto-Routing Software Configuration Guide
Figure 186: Configure Auto-Routing Section
2
3
4
2. Select whether or not automatic routes are generated for the endpoint when auto-routing is enabled
in the Auto-routable drop-down menu.
Table 288: Auto-routable Parameters
Enable Automatic routes allowing incoming and outgoing calls to and from
the endpoint are added to the Route Table when auto-routing is
enabled.
Disable Automatic route generation is turned off for this endpoint.
HardwareDependent Automatic routes are generated if the endpoint belongs to an FXS
interface.
3. Select the SIP gateways to use as the destination of outgoing calls and the source of incoming calls
when generating auto-routing rules in the Auto-routing Gateway drop-down menu.
If you leave the field blank, it is the same as disabling the auto-routing feature.
More than one SIP gateway can be defined. The SIP gateways names are separated by comas.
Example:
gw1,gw2,gw3
When one SIP gateway is defined:
• A route is automatically created from the SIP gateway to the telephony interface.
• A route is automatically created from the telephony interface to the SIP gateway if the
Auto-routing Destination field is empty. Otherwise, the destination of the route uses the
destination defined in the Auto-routing Destination field.
When several SIP gateways are defined:
• Routes are automatically created from each defined SIP gateway to the telephony
interface.
• A route is automatically created from the telephony interface to the destination defined
in the Auto-routing Destination field. No route is created if the destination is left empty.
If available, two additional parameters are displayed:
• If an endpoint has a telephone number that is associated with it, it is displayed in the
corresponding E164 column. This is the User Name field as configured in the SIP >
Registration page as long as the name follows the E.164 syntax.
• If an endpoint has a friendly name that is associated with it, it is displayed in the
corresponding Name column. This is the Friendly Name field as configured in the SIP >
Registration page.
Pleas note that routes are created only if a user name is associated with the telephony endpoint in
the registration table. See “Endpoints Registration” on page 289 for more details.

4. Set the destination to use for the routes from the telephony interface in the Auto-routing Destination
field.
The destination can be:
• route-name: The route destination is set to the route name.
• hunt-name: The route destination is set to the hunt name.
• sip-name: The route destination is set to the SIP interface name.
• isdn-name: The route destination is set to the ISDN interface name.
• r2-name: The route destination is set to the R2 interface name.
• e&m-name: The route destination is set to the E&M interface name.
• fxs-name: The route destination is set to the FXS interface name.
• fxo-name: The route destination is set to the FXO interface name.
5. You can copy the configuration of the selected endpoint to one or more endpoints of the Aastra unit
in the Apply to the Following Endpoints section at the bottom of the page. You can select specific
endpoints by checking them, as well as use the Check All or Uncheck All buttons.
6. When you are finished, you have the choice to:
• Click the Submit button to enable auto-routing.
The current routes applied are displayed in the Call Router > Status web page. They are
added at the end of the routes that are already present, if any. This ensures that the
user-defined routes always have precedence over the automatic routes when both types
of routes apply to the same endpoint.
• Click the Submit & Create Hunt button to perform a submit action and go to the hunt
creation page. This option is available only if the destination is set to an unexisting
hunt.
• Click the Submit & Edit Hunt button to perform a submit action and go to the hunt
edition page. This option is available only if the destination is set to an existing hunt.
• Click the Submit & Create Route button to perform a submit action and go to the route
creation page. This option is available only if the destination is set to an unexisting
route.
• Click the Submit & Edit Route button to perform a submit action and go to the route
edition page. This option is available only if the destination is set to an existing route.
Manual Routing
Auto-routing can only be used if the username of the endpoint is an E.164 string and the username part of the
request-URI of the received INVITE can be converted into an E.164.
The conversion of a username into an E.164 follows these rules:
 The prefix “+” is removed. Note that if the Map Plus To TON International drop-down menu is
set to Enable, the call property ‘type of number’ is set to ‘international’. See “Misc Interop” on
 The visual separator “-” is removed.
 The username parameter is removed. The username parameter is a suffix beginning with “;”.
 All remaining characters need to be “0123456789*#abcdABCD”.
Examples of conversion:
5551234 --> 5551234
#20 --> #20
555-1234 --> 5551234
+1-819-555-1234 --> 18195551234
5551234;parameter --> 5551234
5551234_parameter --> cannot convert
To use a username not compatible with E.164, you must disable the auto-routing and use manual routes.
Figure 187 gives an example of manual routes for an endpoint using “5550001_paramter” as user.

Auto-Routing Software Configuration Guide
Figure 187: Manual Routes Example


Management Parameters
C H A P T E R
40 Configuration Script
This chapter describes the configuration script download feature, which allows updating the Aastra unit
configuration by transferring a configuration script from a remote server or from the local file system. The
Aastra unit is the session initiator, which allows NAT traversal. You can also configure the Aastra unit to
automatically update its configuration.
You can also generate a configuration script from the running configuration of the Aastra unit.
Configuration scripts are files containing textual commands that are sent over the network to a Aastra unit.
Upon receiving the file, the unit executes each command line in sequence. Script commands can assign
values to configuration variables, or execute configuration commands. See “Creating a Configuration Script”
on page 414 for more details on how to create a configuration script.
Scripts are written by the system administrator and can be used to accomplish various tasks, such as
automating recurrent configuration tasks or batch-applying configuration settings to multiple devices. Scripts
can be executed once or periodically at a specified interval. They can also be scheduled to execute when the
Aastra unit restarts.
 Configuration script server setup.
 Configuration script server parameters.
 Configuration download procedure.
 Generating a configuration script from the running configuration.
 Automatic configuration update parameters.
 How to create a configuration script from scratch.
Standards Supported • RFC 959: File Transfer Protocol (client-side only)

• RFC 1350: The TFTP Protocol (Revision 2) (client-side only)
side only)
• RFC 2617: HTTP Authentication: Basic and Digest Access
Authentication
• RFC 3617: Uniform Resource Identifier (URI) Scheme for the
Trivial File Transfer Protocol
• draft-ietf-http-authentication-03
Configuration Script Server
To download a configuration script, you may need to setup the following applications on your computer:
 SNTP server properly configured
 HTTPS server with proper root path

When you perform a configuration script download by using the TFTP (Trivial File Transfer Protocol) protocol,
you must install a TFTP server running on the PC designated as the TFTP server host. It is assumed that you
know how to set the TFTP root path. If not, refer to your TFTP server’s documentation.

Chapter 40 - Configuration Script Configuration Script Server
Configuring the SNTP Server

When you use the automatic configuration script update feature (see “Automatic Configuration Update” on
page 410 for more details) or the HTTPS protocol, you need to have a time server SNTP that is accessible
and properly configured. It is assumed that you know how to configure your SNTP server. If not, refer to your
SNTP server’s documentation. You can also refer to “SNTP Configuration” on page 93 for more details on how
to configure the Aastra unit for a SNTP server.
time.

When you to perform a configuration script download by using the HTTP protocol, you must install a HTTP
server running on the PC designated as the server host. It is assumed that you know how to set the root path.
If not, refer to your HTTP server’s documentation.
Configuring the HTTPS Server

• RFC 2459: X.509 Digital Certificates
• RFC 2818: HTTP Over TLS (client side only)
• RFC 3268: Advanced Encryption Standard (AES)
Ciphersuites for Transport Layer Security (TLS)
• RFC 3280: Internet X.509 Public Key Infrastructure Certificate
and Certificate Revocation List (CRL) Profile
When you perform a configuration script download that requires authentication or privacy by using the HTTP
over the Transport Layer Security (TLS) protocol (HTTPS), you must install a HTTPS server running on the
PC designated as the server host. It is assumed that you know how to set the root path and SSL/TLS security
configuration. If not, refer to your HTTPS server’s documentation.
Caution: You must have a time server SNTP that is accessible and properly configured, or the automatic
configuration update feature may not work properly. It is assumed that you know how to configure your
SNTP server. If not, refer to your SNTP server’s documentation. You can also refer to “SNTP Configuration”
on page 93 for more details on how to configure the Aastra unit for a SNTP server.
When two peers establish a HTTPS connection, they negotiate and decide on a cipher suite to use for data
encryption. The client suggests a list of cipher suites and the server selects one that it supports. Some cipher
suites are more secured than others. The Aastra unit acts as a client.
The Aastra unit suggests a wide range of cypher suites, which includes cipher suites that are not very secure.
The final choice rests with the server and it is thus possible that the transfer uses a SSL/TLS link that is not
very secure.
Aastra recommends to use cipher suites based on the RSA key exchange mechanism, because the Diffie-
Hellman key exchange mechanism introduces a noticeable delay in the HTTPS session establishment.
Furthermore, Aastra recommends using cipher suites based on the following SSL/TLS algorithms:
Table 289: Suggested Secure Parameters
Suggested Parameter Description
Key Exchange Mechanism • RSA

• Diffie-Hellman
Ciphers • AES (128 and 256 bits)
• 3DES (168 bits)

Configuration Script Server Software Configuration Guide
Table 289: Suggested Secure Parameters (Continued)
Message Digests • SHA-1
The following six recommended cipher suites are based on the algorithms of Table 289:
Table 290: Recommended Cipher Suites
ID Name
0x0035 TLS_RSA_WITH_AES_256_CBC_SHA
0x0039 TLS_DHE_RSA_WITH_AES_256_CBC_SHA
0x000a TLS_RSA_WITH_3DES_EDE_CBC_SHA
0x0016 TLS_DHE_RSA_WITH_3DES_EDE_CBC_SHA
0x002f TLS_RSA_WITH_AES_128_CBC_SHA
Certificates
The Aastra unit contains embedded security certificates formatted as per ITU x.509 and RFC 3280. The
certificates are factory-installed. You can also add new certificates as described in “Chapter 46 - Certificates
Management” on page 557.
When contacting a HTTPS server, the Aastra unit establishes a TLS connection by (among others):
 negotiating cipher suites
 checking the server certificates validity (dates)
The Aastra unit then checks the server’s identity by validating the host name used to contact it against the
information found in the server’s certificate, as described in RFC 2818, section 3.1.
If any of the above does not succeed, the Aastra unit refuses the secure connection. To help detect such
errors, you can increase the syslog messages level.

Chapter 40 - Configuration Script Generating a Configuration Script from the Running Configuration
Generating a Configuration Script from the Running Configuration
You can generate a configuration script from the running configuration of the Aastra unit and export it.
You can export the configuration in two ways:
 To a URL you specify with one of the supported transfer protocols.
 By directly downloading the exported script via your web browser. This option uses the
protection provided by your web browser (it is protected if you log on to the unit via HTTPS).
Exporting a Configuration to a URL

The Export Script section allows you to generate a configuration script from the running configuration of the
Aastra unit and export it.
 To export a configuration to a URL:

1. In the web interface, click the Management link, then the Configuration Scripts sub-link.
Figure 188: Management – Configuration Scripts Web Page
2
3
4
5
2. Select the content to export in the generated configuration script in the Content drop-down menu.
Table 291: Exported Configuration Script Content
All Config Exports everything.

Modified Config Export only the configuration that has been modified (differs from the default
values).
3. Set the Service Name field with the name of the service from which to export configuration.
You can use the Suggestion drop-down menu to select one of the available services. You can use
the special value All to export the configuration of all services.
4. Set the Send To URL field with the URL where to send the exported configuration script.
The URL should follow this format:
protocol://[user[:password]@]hostname[:port]/[path/]filename
The brackets [ ] denote an optional parameter.
The filename may contain a %mac% macro that is substituted by the MAC address of the unit at the
moment of sending the configuration script. For instance, the “%mac%.cfg” value for a Aastra unit
with MAC address “0090f12345ab” will be “0090f12345ab.xml”.
The filename may contain macros that are substituted at the moment of sending the configuration
script. The supported macros are:

Generating a Configuration Script from the Running Configuration Software Configuration Guide
• %mac% - the MAC address of the unit

• %version% - the MFP version of the unit
For instance, the “%mac%.cfg” value for a Aastra unit with MAC address “0090f12345ab” will be
“0090f12345ab.xml”.
The transfer protocols supported are:
• TFTP
• FTP
• FILE
Examples of valid URLs:
• tftp://tftpserver.com:69/folder/script.cfg
• ftp://[email protected]/script.cfg
• ftp://username:[email protected]/script.cfg
• file://script.cfg
The protocol’s default port is used if none is specified.
5. Set the Privacy Key field with the key used to encrypt the configuration script to export.
Caution: The Privacy Key field is not accessible if you have the User or Observer access right. See “Users”
The key is encoded in hexadecimal notation. You can thus use characters in the range 0-9, A-F,
and a-f. All other characters are not supported.
The maximum key length is 64 characters, which gives a binary key of 32 bytes (256 bits). It is the
maximum key size supported by the MxCryptFile application.
For instance, a 32-bit key could look like the following: A36CB299.
If the field is empty, the configuration script is not encrypted.
To decrypt the exported configuration script, you must use the MxCryptFile application. MxCryptFile
is a command line tool that encrypts or decrypts files to be exchanged with the Aastra unit. Contact
your sales representative for more details.
6. Initiate the configuration scripts exportation by clicking the Submit & Export Now button at the
bottom of the page.
The Aastra unit immediately generates and transfers a configuration script based on the export
settings set in the previous steps.
Exporting a Configuration Script to your PC

This section describes how to export the configuration of a Aastra unit to the PC.
If you are currently using an unsecure HTTP access, script transfers through web browser are disabled. This
is to avoid transferring the configuration in clear text. To enable the section, you can:
 Access the secure site (recommended) by clicking the corresponding link at the top of the
window. This is the recommended way to proceed.
 Activate unsecure certificate transfer by clicking the corresponding link at the top of the window.
This is not recommended.

Chapter 40 - Configuration Script Generating a Configuration Script from the Running Configuration
 To export a configuration script to the PC:

1. In the Transfer Scripts Through Web Browser section of the Configuration Scripts page, select the
content to export in the generated configuration script in the Content drop-down menu.
Table 292: Exported Configuration Script Content
All Config Exports everything.

Modified Config Export only the configuration that has been modified (differs from the default
values).
Figure 189: Transfer Scripts Through Web Browser Section
3
1
2. If required, set the Privacy Key field with the key used to encrypt the configuration script to export.
The maximum key length is 64 characters, which gives a binary key of 32 bytes (256 bits). It is the
maximum key size supported by the MxCryptFile application.
If the field is empty, the configuration script is not encrypted.
To decrypt the exported configuration script, you must use the MxCryptFile application. MxCryptFile
is a command line tool that encrypts or decrypts files to be exchanged with the Aastra unit. Contact
your sales representative for more details.
3. Click the Export & Download button.

Executing Configuration Scripts Settings Software Configuration Guide
Executing Configuration Scripts Settings
You can execute the configuration in two ways:

 From the configuration script server URL that you specify with one of the supported transfer
protocols.
 By directly executing the script via your web browser. This option uses the protection provided
by your web browser (it is protected if you log on to the unit via HTTPS).
Executing a Script from the Configuration Script Server

This section describes how to configure the IP address and port number of the configuration script server. This
server contains the configuration scripts the Aastra unit will download.
When performing a configuration script download, you can download two different scripts:
 A generic configuration script that should be used to update a large number of units with the
same configuration.
 A specific configuration script that contains the configuration for a single unit, for instance the
telephone numbers of its endpoints.
You can use a specific configuration script but no generic configuration script or vice-versa. You can also use
both generic and specific configuration scripts. When both the generic and specific configuration scripts are
downloaded, settings from the specific configuration script always override the settings from the generic
configuration script. These scripts must be located in the same directory.
Each script is executed independently from the other one. A script that is empty, cannot be found or has an
invalid syntax does not prevent the execution of the other script. If one or both scripts fail, error messages are
sent.
 To set the configuration scripts server parameters:

1. In the Configuration Scripts section of the Configuration Scripts page, set the name of the generic
configuration script to download in the Generic File Name field.
This script should be used to update a large number of units with the same configuration. The script
name is case sensitive hence it must be entered properly.
If you select File in the Transfer Protocol drop-down menu (Step 4), this means that you can select
a script located in the unit’s persistent file system. You can use the Suggestion drop-down menu to
select one of the available scripts in the file system.
To see the content of the unit’s file system persistent memory, go to the File Manager (“Chapter 50
- File Manager” on page 597). All installed configuration scripts/images are listed.
This field may contain some macros that are substituted by the actual value at the moment of
fetching the configuration script. The supported macros are:
• %product% - the Product name of the unit.
• %productseries% - the Product series name of the unit.
For instance, the “%mac%.xml” value for a Aastra unit with MAC address “0090f12345ab” will be
“0090f12345ab.xml”.
If you leave the field empty, the Aastra unit does not download the generic configuration script.

Chapter 40 - Configuration Script Executing Configuration Scripts Settings
Figure 190: Execute Scripts Section
1
2
3
4
5
6
7
8
2. Set the name of the specific configuration script to download in the Specific File Name field.
This script should be used to update the configuration of a single unit. The script name is case
sensitive hence it must be entered properly.
a script located in the unit’s persistent file system. You can use the Suggestion drop-down menu to
select one of the available scripts in the file system.
This field may contain a macro that is substituted by the actual value when downloading the
configuration script. The Aastra unit supports the %mac% macro, which will be substituted by the
MAC address of the unit. For instance, the “%mac%.xml” value for a Aastra unit with MAC address
“0090f12345ab” will be “0090f12345ab.xml”.
This field may contain some macros that are substituted by the actual value when downloading the
configuration script. The supported macros are:
“0090f12345ab.xml”.
If the variable is empty (after macro substitution), the Aastra unit does not download the specific
configuration script.
3. Set the path of the directory where the configuration scripts are located in the Location field.
The path is case sensitive hence it must be entered properly. It is relative to the root of the
configuration scripts server. Use the “/” character when defining the path to indicate sub-directories.
“0090f12345ab.xml”.
The path differs depending on the transfer protocol selected (see Step 5).
Example: All Transfer Protocols Except File
Let’s consider the following example for all protocols except File:
• The directory that contains the configuration script is called: Config_Script.

• This directory is under C:/Root/Download.

Table 293: Path Configurations Example
Root Path Corresponding Path Name
c:/root/download Config_Script
c:/ root/download/Config_Script
c:/root download/Config_Script
The following are some tips to help your download process:

• Use the “/” character when defining the path to indicate sub-directories. For instance,
root/download.
• If you are using the TFTP protocol to download the software, note that some TFTP
servers on Windows do not recognize the “/” character and produce an error. In this
case, use the “\” character.
• Use basic directory names, without spaces or special characters such as “~”, “@”, etc.,
which may cause problems.
• Cut and paste the path and/or name of the directory that contains the extracted scripts
into the configuration download path of the Aastra unit (you may have to convert “\” into
“/”) to eliminate typographical errors.
Note that you can define the C:/Root/Download part as you want. The script names may also differ
from the example shown above.
When the Transfer Protocol is set to File, you may prefix the path by one of the following to indicate
storage media:
• Persistent: for onboard persistent storage. The configuration script is saved into the
persistent file system of the Aastra unit (in flash memory). This is the default value.
• Volatile: for onboard non-persistent storage. The configuration script is saved into the
non-persistent RAM memory of the Aastra unit. All information is lost the next time the
unit restarts.
Table 294: Path Configurations Example (File)
Location Corresponding Path Name
Onboard persistent storage of the Persistent:Script-1

Aastra unit under the directory or
“Script-1”
Script-1
4. Set the transfer protocol to transfer the configuration scripts in the Transfer Protocol field.
You can select from five different transfer protocols:
• HTTPS: HyperText Transfer Protocol over Transport Layer Security.
• FTP: File Transfer Protocol. Note that the Aastra unit FTP client does not support the
EPSV command.
Note: The configuration script download via TFTP can only traverse NATs of types “Full Cone” or
“Restricted Cone”. If the NAT you are using is of type “Port Restricted Cone” or “Symmetric”, the script
transfer will not work.
• File: Complete path to a configuration image in a storage device. You can view and
manage all files created with the File transfer protocol by using the File Manager. See
“File Manager” on page 597 for more details.
HTTP and HTTPS support basic or digest authentication mode as described in RFC 2617. HTTPS
requires a valid certificate.

Chapter 40 - Configuration Script Executing Configuration Scripts Settings
If you have selected HTTP or HTTPS, please note that your server may activate some caching
mechanism for the script download. This mechanism caches the initial script download for later
processing, thus preventing changes or update of the original script. This can cause strange
problems if you want to edit a configuration script to modify values and upload it immediately. The
result will still return the original script and not the new one.
5. If your server requires authentication when downloading the configuration script, set the following:
6. Set the static configuration scripts server IP address or domain name and port number in the Host
Name field.
This is the current address of the PC that hosts the configuration scripts.
Use the special port value zero to indicate the protocol default. For instance, the TFTP default port
is 69, the HTTP default port is 80, and the HTTPS default port is 443.
The default value is 0.0.0.0:0.
7. Set the key used to decrypt configuration scripts when they are encrypted in the Privacy key field.
You can secure the exchange of configuration scripts between the server and the Aastra unit. A
privacy key allows the unit to decrypt a previously encrypted configuration script.
To encrypt a configuration script, you must use the MxCryptFile application. MxCryptFile is a
command line tool that encrypts files before sending them to the Aastra unit. Contact your sales
representative for more details.
Each character encodes 4 bits and the maximum key length is 112 characters, which gives a binary
key of 56 bytes. It is the maximum supported by the MxCryptFile application.
This key must match the key used for the encryption of the relevant configuration script.
If the field is empty, the configuration script is not decrypted by the unit and the configuration update
fails.
Encryption is auto-detected.
8. Define whether or not to allow the execution of a script even if it is identical to the last executed script
in the Allow Repeated Execution drop-down menu.
Table 295: Allow Repeated Execution Parameters
Auto Uses the configured value of ScriptsAllowRepeatedExecution.

Enable Allows repeated execution of the same script.
Disable Does not allow repeated execution of the same script.
The script retry mechanism is not enabled for the DHCP triggered scripts (see “DHCPv4 Auto-
Provisioning” on page 412 for more details).

• Click Submit if you do not need to set other parameters.

• Click Submit & Execute Now to execute the script.
Executing a Script from your PC

This section describes how to execute a script located on the PC.
If you are currently using an unsecure HTTP access, script transfers through web browser are disabled. This
is to avoid transferring the configuration in clear text. To enable the section, you can:
 Access the secure site (recommended) by clicking the corresponding link at the top of the
window. This is the recommended way to proceed.
 Activate unsecure certificate transfer by clicking the corresponding link at the top of the window.
This is not recommended.
 To execute a script from the PC:

1. In the Transfer Scripts Through Web Browser section of the Configuration Scripts page, type the
name of a script in the Upload Parameters field or select an existing one on the PC with the Browse
button.
When a script is executed, it is not installed in the unit’s file system persistent memory. You can click
the Clear Selection link to empty the field and enter another name.
Figure 191: Transfer Scripts Through Web Browser Section
3
1
2. If required, set the key used to decrypt configuration scripts when they are encrypted in the Privacy
key field.
You can secure the exchange of configuration scripts between the server and the Aastra unit. A
privacy key allows the unit to decrypt a previously encrypted configuration script.
To encrypt a configuration script, you must use the MxCryptFile application. MxCryptFile is a
command line tool that encrypts files before sending them to the Aastra unit. Contact your sales
representative for more details.
Each character encodes 4 bits and the maximum key length is 112 characters, which gives a binary
key of 56 bytes. It is the maximum supported by the MxCryptFile application.
This key must match the key used for the encryption of the relevant configuration script.
If the field is empty, the configuration script is not decrypted by the unit and the configuration update
fails.
Encryption is auto-detected.
3. Click the Upload & Execute button.

Chapter 40 - Configuration Script Configuration Download Procedure
Configuration Download Procedure
The following steps explain how to download configuration scripts from the web interface.
Note: The configuration download via TFTP can only traverse NATs of types “Full Cone” or “Restricted
Cone”. If the NAT you are using is of type “Port Restricted Cone” or “Symmetric”, the file transfer will not
work.
 To download configuration scripts:

1. Place the configuration scripts to download on the computer hosting the configuration scripts server.
These scripts must be in a directory under the server’s root path.
2. Initiate the configuration scripts download by clicking the Submit & Execute Now button at the
bottom of the page.
The Aastra unit immediately downloads the configuration scripts.
Automatic Configuration Update
This section describes how to configure the Aastra unit to automatically update its configuration. This update
can be done:
 Every time the Aastra unit restarts.
 At a specific time interval you can define.
Automatic Update on Restart

The Aastra unit may download new configuration scripts each time it restarts.
NAT Variations
NAT treatment of UDP varies among implementations. The four treatments are:
• Full Cone: All requests from the same internal IP address and port are mapped to the same
external IP address and port. Furthermore, any external host can send a packet to the internal
host by sending a packet to the mapped external address.
• Restricted Cone: All requests from the same internal IP address and port are mapped to the
same external IP address and port. Unlike a full cone NAT, an external host (with IP address
X) can send a packet to the internal host only if the internal host had previously sent a packet
to IP address X.
• Port Restricted Cone: Similar to a restricted cone NAT, but the restriction includes port
numbers. Specifically, an external host can send a packet, with source IP address X and
source port P, to the internal host only if the internal host had previously sent a packet to IP
address X and port P.
• Symmetric: All requests from the same internal IP address and port, to a specific destination
IP address and port, are mapped to the same external IP address and port. If the same host
sends a packet with the same source address and port, but to a different destination, a
different mapping is used. Furthermore, only the external host that receives a packet can send
a UDP packet back to the internal host.
For more details on NAT treatments, refer to RFC 3489.

Automatic Configuration Update Software Configuration Guide
 To set the automatic update every time the Aastra unit restarts:
1. Set the configuration scripts parameters as defined in “Executing Configuration Scripts Settings” on
page 405.
These scripts must be in a directory under the root path.
3. In the Automatic Script Execution section of the Configuration Scripts page, select Enable in the
Execute on Startup drop-down menu.
Figure 192: Automatically Update Scripts Section
The automatic configuration update will be performed each time the Aastra unit restarts.
The unit configuration is only updated if at least one parameter value defined in the downloaded
configuration scripts is different from the actual unit configuration.
Automatic Update at a Specific Time Interval

You can configure the Aastra unit to download new configuration scripts at a specific day and/or time.
 To set the automatic update at a specific time interval:

1. Set the configuration scripts parameters as defined in “Executing Configuration Scripts Settings” on
page 405.
These scripts must be in a directory under the root path.
3. In the Automatic Script Execution section of the Configuration Scripts page, select Enable in the
Execute Periodically drop-down menu.
3
4
5
6
4. Select the time base for configuration updates in the Time Unit drop-down menu.
Table 296: Time Unit Parameters
Minutes Updates the unit’s configuration every x minutes.

Hours Updates the unit’s configuration every x hours.
Days Updates the unit’s configuration every x days. You can define the time
of day when to perform the update in the Time of Day field (see Step 6).
You can specify the x value in the Period field (see Step 5).
5. Set the waiting period between each configuration update in the Period field.

Chapter 40 - Configuration Script Automatic Configuration Update
Available values are from 1 to 60. The time unit for the period is specified in the Time Unit field (see
Step 4).
6. If you have selected Days in Step 4, set the time of the day when to initiate a configuration update
in the Time of Day field.
The time of the day is based on the Static Time Zone field of the Network - Host page (see “Time
Configuration” on page 94 for more details).
You must have a time server SNTP that is accessible and properly configured or the automatic
configuration update feature may not work properly. It is assumed that you know how to configure
your SNTP server. If not, refer to your SNTP server’s documentation.
time.
The configuration scripts are downloaded at the first occurrence of this value and thereafter at the
period defined by the Period field. Let’s say for instance the automatic unit configuration update is
set with the time of day at 14h00 and the update period at every 2 days.
• If the automatic update is enabled before 14h00, the first update will take place the
same day at 14h00, the second update two days later at the same hour, and so on.
• If the automatic update is enabled after 14h00, the first update will take place the day
after at 14h00, the second download two days later at the same hour, and so on.
Available values are -1, and from 0 to 23.
When setting the variable to -1, the time of the day at which the Aastra unit first downloads the
configuration scripts is randomly selected.
DHCPv4 Auto-Provisioning
Note: This feature does not support IPv6. See “IPv4 vs. IPv6 Availability” on page 85 for more details.
You can configure the Aastra unit to automatically download new configuration scripts upon receiving options
66 (tftp-server) or 67 (bootfile) in a DHCPv4 answer. A DHCP answer includes both Bound and Renew.
The contents of the option 66 or 67 defines which script to download. The unit’s configuration is not used to
download the script. This allows the unit, for instance, to download a script from a server after a factory reset
and to reconfigure itself without a specific profile.
The syntax of options 66 and 67 is as follows:
[FileType] = [protocol]://[username] :[password]@[fqdn server]/[path]
For instance:
Script=https://admin :[email protected]/Mx3000config/%mac%.cfg
The Aastra unit supports only the Script file type for now.
The following is an example of a valid option 67 (Bootfile):
Option: (t=67, l=53) Bootfile name = "Script=http://192.168.50.1/digest/
%mac%__2.0.6.84.cfg"
Option: (67) Bootfile name
Length: 53
Value: 5363726970743D687474703A2F2F3139322E31136382E3530...

Automatic Configuration Update Software Configuration Guide
 To set DHCPv4 auto-provisioning:

1. In the Automatic Script Execution section of the Configuration Scripts page, set the Allow DHCP to
Trigger Scripts Execution drop-down menu with the proper behaviour.
When enabled, the DHCPv4 options tftp-server (option 66) and bootfile (option 67) are used to
download a configuration script. If this configuration script is identical to the last executed script, it
will not be run again. The script retry mechanism is not enabled for the DHCPv4 triggered scripts
(see “Executing Configuration Scripts Settings” on page 405 for more details).
If the two options are received, both scripts are executed independently. The script defined by the
tftp-server (option 66) option is executed first.
If you are using HTTPS to transfer scripts, you must have a time server SNTP that is accessible and
properly configured. It is assumed that you know how to configure your SNTP server. If not, refer to
your SNTP server’s documentation.
time.
When a DHCPv4 download script is configured in HTTPS, the script execution is deferred and a 30
seconds timer is started to let enough time for the NTP synchronization.
This timer is independent for each HTTPS script launched. If, for instance, a DHCPv4 answer has
both option 66 and 67 configured in HTTPS and if the Update on Restart feature is used, up to 1
min 30 seconds can pass before any other operation such as backup, restore, script execution can
be processed.
Once the NTP synchronization is established, the deferred scripts are started immediately one after
the other, ending the timer.
When synchronization is already established, there is no timer, even in HTTPS.
Number of Retries
When using the automatic configuration update (on restart or at a specific interval), the Aastra unit may
encounter a problem upon restarting the unit (such as a DHCP server problem) that prevents the update to
succeed. You can define a maximum number of attempts to retry a script transfer until it succeeds when it fails
upon an automatic transfer on restart or automatic periodic transfer. The retries are only attempted if the server
is unreachable. Unreachable port or file not found errors don't trigger the retry mechanism.The time interval
between each retry is 30 seconds.
 To set the number of retries:

1. In the confMIB, set the number of retries in the scriptsTransferRetriesNumber variable.
conf.scriptsTransferRetriesNumber="Value"

Chapter 40 - Configuration Script Creating a Configuration Script
where Value may be as follows

• -1 means a retry to infinity.
• 0 means no retry.
The maximum number of retries is 100.
Creating a Configuration Script
Configuration scripts are text files that contain command lines interpreted by the Aastra unit. Most commands
contained in a script assign values to configuration variables. Script commands can also execute configuration
commands. This configuration script can then be downloaded into the Aastra unit as described in the current
chapter.
Writing configuration scripts requires a bit of knowledge about the Aastra unit’s configuration variables tree
structure. Each parameter that is accessed via the unit's web interface maps to a variable in the configuration
tree. For detailed information on these mappings, please refer to “Appendix D - Web Interface – SNMP
Variables Mapping” on page 641.
Configuration scripts use the Aastra proprietary scripting language, as described in “Appendix B - Scripting
Language” on page 627.
Refer to “Appendix B - Scripting Language” on page 627 for samples of configurations you can use in a
configuration script. The samples include the configuration required to perform a basic call between an ISDN
telephone and an analog telephone. These samples may also be used in the Aastra unit Command Line
Interface.

C H A P T E R
41 Configuration Backup/Restore
This chapter describes the configuration backup/restore feature, which allows you to backup (upload) all the
SNMP (MIB) and Web configuration of the Aastra unit into a configuration image file located on a remote
server or to the local file system.
 Configuration backup download server setup.
 Backup/restore configuration parameters.

side only)
Authentication
Configuration Backup Download Server
To backup/restore a configuration image, you may need to setup the following applications on your computer:

When you perform a configuration backup/restore by using the TFTP (Trivial File Transfer Protocol) protocol,
you must install a TFTP server running on the PC designated as the TFTP server host. It is assumed that you
know how to set the TFTP root path. If not, refer to your TFTP server’s documentation.

When you use the HTTPS protocol, you need to have a time server SNTP that is accessible and properly
configured. It is assumed that you know how to configure your SNTP server. If not, refer to your SNTP server’s
documentation. You can also refer to “SNTP Configuration” on page 93 for more details on how to configure
the Aastra unit for a SNTP server.
time.

Chapter 41 - Configuration Backup/Restore Configuration Backup Download Server

When you to perform a configuration backup/restore by using the HTTP protocol, you must install a HTTP
server running on the PC designated as the server host. It is assumed that you know how to set the root path.
If not, refer to your HTTP server’s documentation.

When you perform a configuration backup/restore that requires authentication or privacy by using the HTTP
over the Transport Layer Security (TLS) protocol (HTTPS), you must install a HTTPS server running on the
PC designated as the server host. It is assumed that you know how to set the root path and SSL/TLS security
configuration. If not, refer to your HTTPS server’s documentation.
very secure.

• Diffie-Hellman
• 3DES (168 bits)
ID Name
Certificates

Backup/Restore Configuration Software Configuration Guide
Backup/Restore Configuration
This section describes how to set the backup/restore configuration parameters and some related files (e.g.
certificates). You can restore this configuration in case the Aastra unit loses it for any reason or to clone a unit
with the configuration of another unit. The configuration backup images are in XML format and may be
encrypted or in clear text.
Note: The files under the File service are not included in the backup process. In the same way, the restore
process will not remove any file under the File service.
Please note that you can use a backup file from an older firmware version and use it in a unit with a more
recent firmware version. However, a backup file from a newer firmware version than the one actually in the
unit cannot be used for a restore operation on the unit. For instance, let’s say you perform the following
backups:
 In firmware v1.1r4.1, you make the backup “Backup_X1”.
 In firmware v1.1r4.2 you make the backup “Backup_X2”
Application v1.1r4.2 is more recent than v1.1r4.1. The following table describes the various scenarios
possible.
Table 299: Backup Matrix
Not
Scenario Supported
Supported
Apply the backup “Backup_X1” in a unit with firmware v1.1r4.1.

You can backup or restore the configuration to/from two sources:

 To/from an image located on an image server
 To/from an image located on your PC (transfer images through the web browser)
This section explains both methods.

Chapter 41 - Configuration Backup/Restore Backup/Restore Configuration
Performing a Backup/Restore to/from on an Image Server or File System

 To set the configuration backup/restore parameters:
1. In the web interface, click the Management link, then the Backup / Restore sub-link.
Figure 195: Management – Backup / Restore Web Page
2
3
4
5
6
8
9
2. Set the name of the configuration image in which you want to backup or from which you want to
restore the Aastra unit configuration in the File Name field.
The file name is case sensitive hence it must be entered properly. Make sure to write the file
extension.
an image located in the unit’s persistent file system. You can use the Suggestion drop-down menu
to select one of the available images in the file system.
This field may contain a macro that is substituted by the actual value when backing up or restoring
the unit's configuration. The Aastra unit supports the %mac% macro, which will be substituted by
the MAC address of the unit. For instance, the “%mac%.bkp” value for a Aastra unit with MAC
address “0090F12345AB” will be “0090F12345AB.bkp”.
This field may contain macros that are substituted by the actual value when backing up or restoring
the unit's configuration. The supported macros are:
• %mac% - the MAC address of the unit.
• %version% - the MFP version of the unit.
For instance, the “%mac%.bkp” value for a Aastra unit with MAC address “0090F12345AB” will be
“0090F12345AB.bkp”.

3. Select a transfer protocol to transfer a configuration image in the Transfer Protocol drop-down
menu.
You can select from five different transfer protocols:
EPSV command.
• File: Complete path to a configuration image in the Aastra unit’s onboard storage
space. You can view and manage all files created with the File transfer protocol by
using the File Manager. See “File Manager” on page 597 for more details.
Note: The configuration image backup via TFTP can only traverse NATs of types “Full Cone” or “Restricted
Cone”. If the NAT you are using is of type “Port Restricted Cone” or “Symmetric”, the transfer will not work.
The backup operation currently supports the following protocols:
• TFTP
• FTP
• File
The restore operation supports all the transfer protocols.
mechanism for the configuration image transfer. This mechanism caches the initial configuration
image transfer for later processing, thus preventing changes or update of the original image. This
can cause strange problems if you want to edit a configuration image to modify values and upload
it immediately. The result will still return the original image and not the new one.
4. Set the configuration backup/restore server hostname or FQDN and IP port in the Host Name field.
This is the current address and port number of the PC that hosts the configuration image file. Use
the special port value 0 to indicate the protocol default. For instance, the TFTP default port is 69
and the HTTP default port is 80.
NAT Variations
NAT treatment of UDP varies among implementations. The four treatments are:
• Full Cone: All requests from the same internal IP address and port are mapped to the same
external IP address and port. Furthermore, any external host can send a packet to the internal
host by sending a packet to the mapped external address.
• Restricted Cone: All requests from the same internal IP address and port are mapped to the
same external IP address and port. Unlike a full cone NAT, an external host (with IP address
X) can send a packet to the internal host only if the internal host had previously sent a packet
to IP address X.
• Port Restricted Cone: Similar to a restricted cone NAT, but the restriction includes port
numbers. Specifically, an external host can send a packet, with source IP address X and
source port P, to the internal host only if the internal host had previously sent a packet to IP
address X and port P.
• Symmetric: All requests from the same internal IP address and port, to a specific destination
IP address and port, are mapped to the same external IP address and port. If the same host
sends a packet with the same source address and port, but to a different destination, a
different mapping is used. Furthermore, only the external host that receives a packet can send
a UDP packet back to the internal host.
For more details on NAT treatments, refer to RFC 3489.

5. Set the path of the directory where the configuration image is located in the Location field.
The path is case sensitive hence it must be entered properly. It is relative to the root of the
configuration transfer server. Use the “/” character when defining the path to indicate sub-
directories.
“0090f12345ab.xml”.
The path differs depending on the transfer protocol selected (see Step 4).
Example: All Transfer Protocols Except File
Let’s consider the following example for all protocols except File:
• The directory that contains the configuration image is called: Config_Image.
c:/root/download Config_Image
c:/ root/download/Config_Image
c:/root download/Config_Image
The following are some tips to help your process:

root/download.
• If you are using the TFTP protocol to download the software, note that some TFTP
servers on Windows do not recognize the “/” character and produce an error. In this
case, use the “\” character.
• Cut and paste the path and/or name of the directory that contains the file into the
configuration download path of the Aastra unit (you may have to convert “\” into “/”) to
eliminate typographical errors.
Note that you can define the C:/Root/Download part as you want. The file name may also differ
Example: Transfer Protocol is File
When the Transfer Protocol is set to File, you may prefix the path by one of the following to indicate
storage media:
• Persistent: for onboard persistent storage. The configuration image is saved into the
persistent file system of the Aastra unit (in flash memory). This is the default value.

• Volatile: for onboard non-persistent storage. The configuration image is saved into the
non-persistent RAM memory of the Aastra unit. All information is lost the next time the
unit restarts.
Table 301: Path Configurations Example (File)
Location Corresponding Path Name
Onboard persistent storage of the Persistent:Backup-1

Aastra unit under the directory or
“Backup-1”
Backup-1
6. If your server requires authentication, set the following:

7. Define the Backup Content drop-down menu with the information to include in the backup.
Table 302: Backup Content Parameters
Config Only the unit's configuration is included in the backup image.

Config And Certificates The unit's configuration and certificates are included in the backup
image. Aastra strongly recommends to activate encryption when
including certificates in the backup image because host certificates
include the private key (see Steps 9-10).
8. Set the privacy algorithm in the Privacy Algorithm field.

This defines the encryption method to use for backup operations. This parameter is not used for
restore operations.
You can secure the exchange of configuration image between the server and the Aastra unit. A
privacy key allows the unit to decrypt a previously encrypted configuration image. During a restore
of the backup image, the encryption is auto-detected.
The configuration image must have been encrypted before use.
Table 303: Privacy Algorithm
None Backup images are not encrypted.

DefaultAlgo Backup images are encrypted with the default algorithm.
9. Set the decryption key in the Privacy key field.
Caution: The Privacy key field is not accessible if you have the User or Observer access right. See “Users”
This is the key used for:

• backup operations to encrypt backup images
• restore operations to decrypt backup images when encrypted (encryption is auto-
detected).

The key is encoded in hexadecimal notation. You can thus use characters in the range 0-9, A-F. All
other characters are not supported.
Each character encodes 4 bits of the key. For instance, a 32-bit key requires 8 characters.
• If you enter too many bits, the key is truncated to the first 448 bits.
• If you do not enter enough bits, the key is padded with zeros.
This key must match the key used for the encryption of the configuration image. If the variable is
empty, the configuration image is not decrypted.
• To save your settings without performing a backup/restore, click Submit.
• To save your settings and perform a backup now, click Submit & Backup Now.
• To save your settings and perform a restore now, click Submit & Restore Now.
Transferring Images Through the Web Browser

This section describes how to perform a backup to or restore from an image located on the PC.
To see the content of the unit’s file system persistent memory, go to the File Manager (“Chapter 50 - File
Manager” on page 597). All installed configuration scripts/images are listed.
 To perform a backup to or restore from an image on the PC:

1. In the Transfer Images Through Web Browser section of the Backup / Restore page, type the name
of an image in the Upload Parameters field or select an existing one on the PC with the Browse
button.
If you are currently using an unsecure HTTP access, the Transfer Images Through Web Browser
section is disabled. This is to avoid transferring an image in clear text. To enable the section, access
the secure site by clicking the Activate unsecure image transfer through web browser link at the top
of the window.
When an image is run, it is not installed in the unit’s file system persistent memory. You can click
the Clear Selection link to empty the field and enter another name.
Figure 196: Transfer Images Through Web Browser Section
3
1
2
2. Set the decryption key in the Privacy key field.

This is the key used for:
• backup operations to encrypt backup images
• restore operations to decrypt backup images when encrypted (encryption is auto-
detected).
The key is encoded in hexadecimal notation. You can thus use characters in the range 0-9, A-F. All
other characters are not supported.
Each character encodes 4 bits of the key. For instance, a 32-bit key requires 8 characters.
• If you enter too many bits, the key is truncated to the first 448 bits.
• If you do not enter enough bits, the key is padded with zeros.
This key must match the key used for the encryption of the configuration image. If the variable is
empty, the configuration image is not decrypted.
3. Click the Upload Now button.

C H A P T E R
42 Firmware Download
This chapter describes how to install, uninstall and update software components on the Aastra unit by using
the web interface, according to a supplied Firmware Pack selection.
Note: If you have backed up the configuration of your unit, Aastra recommends that you perform a new
backup every time you upgrade the firmware pack of the unit to avoid restore issues.

 What is a firmware pack?
 Firmware pack server setup.
 Firmware pack version and name to download.
 Transfer parameters.
 Firmware pack update procedure.

side only)
Authentication
What is a Firmware Pack?
A firmware pack file is a regular zip file that contains the modules and features to install on the Aastra unit.
When unzipping a firmware pack, the contents is extracted according to a pre-defined tree architecture. This
creates a directory that contains the files required for the Aastra unit to properly update its firmware. The
firmware pack contains all the modules to install. When performing the upgrade operation, the Aastra unit
checks the modules versions of the firmware pack against its own modules versions and installs only the
modules that have changed.
Note: The currently installed firmware pack is only required when downgrading.
Before Performing a Firmware Upgrade or Downgrade
To download a firmware pack, you may need to setup the following applications on your computer:
 MIB browser (with the current Aastra unit MIB tree)
 Firmware pack zip file

Chapter 42 - Firmware Download Before Performing a Firmware Upgrade or Downgrade

 Syslog daemon (optional)

When you perform a firmware pack update by using the TFTP Trivial File Transfer Protocol) protocol, you must
install a TFTP (server running on the PC designated as the update files server. This PC must not have a
firewall running. Aastra also recommends to place the PC and the Aastra unit in the same subnet.
It is assumed that you know how to set the TFTP root path. If not, refer to your TFTP server’s documentation.

When you use the HTTPS protocol, you need to have a time server SNTP that is accessible and properly
configured. It is assumed that you know how to configure your SNTP server. If not, refer to your SNTP server’s
documentation. You can also refer to “SNTP Configuration” on page 93 for more details on how to configure
the Aastra unit for a SNTP server.
time.

When you perform a firmware pack update by using the HTTP protocol, you must install a HTTP server running
on the PC designated as the update files server. This PC must not have a firewall running. Aastra also
recommends to place the PC and the Aastra unit in the same subnet.
It is assumed that you know how to set the root path. If not, refer to your HTTP server’s documentation.

When you perform a firmware pack update that requires authentication or privacy by using the HTTP over the
Transport Layer Security (TLS) protocol (HTTPS), you must install a HTTPS server running on the PC
designated as the update files server. It is assumed that you know how to set the root path and set the SSL/
TLS security configuration. If not, refer to your HTTPS server’s documentation.
very secure.

Firmware Packs Configuration Software Configuration Guide

• Diffie-Hellman
• 3DES (168 bits)
ID Name
Certificates
Caution: You must have a time server SNTP that is accessible and properly configured. It is assumed that
you know how to configure your SNTP server. If not, refer to your SNTP server’s documentation. You can
also refer to “SNTP Configuration” on page 93 for more details on how to configure the Aastra unit SNTP
client.
Firmware Packs Configuration
This section allows you to define the firmware pack version and name(s) to properly download them.

Chapter 42 - Firmware Download Firmware Packs Configuration
 To set the firmware pack parameters:

1. In the web interface, click the Management link, then the Firmware Download sub-link.
Figure 197: Management – Firmware Download Web Page
4
5
2. In the Firmware Packs Installed section of the Firmware Upgrade page, click one of the available
buttons if required:
Table 306: Available Buttons
Button Description
Factory Reset You can apply a factory reset to the current unit by clicking the Factory Reset
button. See “Factory Reset” on page 16 for more details.
Rollback You can revert back to the previously installed MFP found in the recovery bank
at any time by clicking the Rollback button. If the recovery bank contains a
MFP that can be used, it is displayed in the Bank column of the Firmware
Packs Installed section. When a rollback is performed, the configuration of the
MFP in the recovery bank applies. The current configuration is lost. The
Rollback button is displayed only if the current bank's application and the
recovery bank's application both support the rollback mechanism and have
been both installed from an application supporting the rollback. Otherwise, the
Rollback button is not displayed.
Note: This feature does not apply to the Aastra TA7102i model.
3. In the Firmware Packs Configuration section of the Firmware Upgrade page, enter the version of
the firmware pack to install in the Version field.
Currently, you cannot install two firmware packs with different versions.
4. Set the Automatic Restart Enable drop-down menu with whether or not to automatically restart the
system when needed for completing a firmware update operation.
You can also set a grace delay in the next step.
5. If automatic restart is enabled, set the Automatic Restart Grace Delay field with the grace delay, in
minutes, that the unit waits for all telephony calls to be terminated before the automatic restart can
occur.
The maximum value is set to 10080 minutes (7 days).
During that delay, it is impossible to make new calls but calls in progress are not terminated. When
all calls are completed, then the unit restarts.

Transfer Configuration Software Configuration Guide
You can also set a services restart grace period as described in “Graceful Restart of Services” on
page 56.
6. Enter the name of up to five firmware packs to install in the Firmware Pack fields.
You can install several firmware packs at the same time. In that case, enter the firmware pack
names in different rows of the table.
When extracting the content of the ZIP file, available firmware packs are listed as directories under
the xxx/FirmwarePacks directory.
Note: The Language drop-down menu currently supports only English.
7. Proceed to “Transfer Configuration” on page 427.
Transfer Configuration
The following describes how to configure the transfer parameters required to perform a firmware update.
 To setup the firmware download path:

1. In the Transfer Configuration section of the Firmware Upgrade page, select a transfer protocol to
transfer a firmware pack in the Transfer Protocol drop-down menu.
Figure 198: Transfer Configuration Section
1
2
3
4

EPSV command.
mechanism for the firmware pack download.
2. Set the static update files server IP address or domain name and port number to use when
downloading a firmware pack in the Host Name field.
This is the current address and port number of the PC that hosts the firmware packs. Use the special
port value 0 to indicate the protocol default. For instance, the TFTP default port is 69, the HTTP
default port is 80, and the HTTPS default port is 443.
This parameter is not required if you have selected the File transfer protocol.
3. Set the firmware download path in the Location field.
This is the location of the folder that contains the modules to download into the Aastra unit. In other
words, this is where the zip file containing the firmware pack has been extracted. This path is
relative to the root of the external media and excludes.
Let’s consider the following example:
• The directory that contains the files required for download is called:

Chapter 42 - Firmware Download Transfer Configuration

c:/root/download N/A
c:/ root/download
c:/root download
The following are some tips to help your download process:

root/download.
• If you are using the TFTP protocol, note that some TFTP servers on Windows do not
recognize the “/” character and produce an error. In this case, use the “\” character.
• Cut and paste the path and/or name of the directory that contains the extracted files
into the firmware download path of the Aastra unit (you may have to convert “\” into “/”)
to eliminate typographical errors.
Note that you can define the C:/Root/Download part as you want. The file names may also differ
4. If your server requires authentication when downloading a firmware pack, set the following:
5. Proceed to “Firmware Pack Update Procedure” on page 429.
Certificate Validation
When downloading a MFP from an HTTPS server, you can define the level of security to use when validating
the server's certificate.
Table 308: Certificate Validation Parameters
NoValidation Allow a connection to the server without validating its certificate. The only condition is to
receive a certificate from the server. This option provides partial security and should be
selected with care.
HostName Allow a connection to the server by validating its certificate is trusted and valid. The
validations performed on the certificate include the expiration date and that the Subject
Alternate Name (SAN) or Common Name (CN) matches the FQDN or IP address of the
server.

Firmware Pack Update Procedure Software Configuration Guide
 To set the certificate validation parameter:

1. In the fpuMIB, set the certificate validation behaviour in the MfpTransferCertificateValidation
variable.
fpu.MfpTransferCertificateValidation="Value"

Table 309: Certificate Validation Values
Value Meaning
100 NoValidation
200 HostName
Firmware Pack Update Procedure
The following describes how to update the firmware pack of the Aastra unit.
Extracting the Firmware Pack Zip File

The firmware pack zip file contains the firmware information required for the update. Extract the contents of
the zip file on the PC designated as the update files server without modifying the defined folder name. This
creates a directory that contains the files required for the Aastra unit to properly update its firmware.
You must extract the zip file under the root path as defined in the update files server or the firmware pack
update will not proceed.
Launching a Firmware Pack Update

The following describes how to launch a firmware pack update.
 To launch the firmware pack update:

1. If not already done, set the firmware packs parameters as defined in “Firmware Packs
Configuration” on page 425.
2. If not already done, unzip the firmware pack file as described in “Extracting the Firmware Pack Zip
File” on page 429.
3. If not already done, set the transfer configuration parameters as described in “Transfer
Configuration” on page 427.
• To save your settings without performing a firmware update, click Submit.
• To save your settings and perform a firmware update now, click Submit & Install Now.
The firmware pack update may take several minutes, depending on your Internet
connection, network conditions and servers conditions.
Caution: Aastra recommends to close and re-open your Web browser after a reboot that installs a firmware
update. This is because your browser may activate a caching mechanism for some files. This mechanism
caches some of the files to improve performance. This may cause problems when the cached files change
in the Aastra unit after a firmware update and the web pages are no longer compatible with the cached files.

Chapter 42 - Firmware Download Firmware Pack Update Status
Firmware Pack Downgrade

It is possible to downgrade a Aastra unit from the current version to an older version. The procedure is the
same as with a firmware upgrade.
Firmware Pack Update Status
When the Aastra unit initiates a firmware pack update, the LEDs indicate the status of the process.
Table 310: LED States in Firmware Pack Update
Event LED State
Firmware pack downloading All LEDs are cycling from left to right, individually blinking 1 Hz, 33%
and writing duty.
Warning: Do not turn the Aastra unit off while in this state.
Firmware pack download failed All LEDs are blinking at 3 Hz, 50% duty. One LED out of two has a 180
degree phase. This pattern lasts for 8 seconds.
You can also view the firmware pack update status in the Status section of the Firmware Upgrade page.
Note: When the firmware pack update fails, the Aastra unit tries to download the firmware three times. In
some cases, the unit may also restart.
Spanning Tree Protocol (STP)
Many network switches use the Spanning Tree Protocol (STP) to manage Ethernet ports activity. When a
firmware pack update occurs, the Ethernet connector of the Aastra unit may switch off. This shutdown may
trigger these network switches to shutdown the matching Ethernet port for at least one minute. This shutdown
on the switch side can prevent firmware pack update.
To prevent this, the Aastra unit supports the STP. However, this management has a potential time cost. It may
appear from time to time that firmware pack updates take more time. This is normal.
When using the unit, Aastra recommends to disable the Spanning Tree Protocol on the network to which the
unit is connected.

C H A P T E R
43 Certificates Management
This chapter describes how to transfer and manage certificates into the Aastra unit.
Standards Supported • RFC 3280: Internet X.509 Public Key Infrastructure Certificate
Introduction
The Aastra unit uses digital certificates, which are a collection of data used to verify the identity of the holder
or sender of the certificate.
The certificates contain the following information:
 certificate name
 issuer and issued to names
 Validity period (the certificate is not valid before or after this period)
 Usage of the certificate (Identifies in which role or context a certificate can be used by the host
it authenticates).
• TlsClient: The certificate identifies a TLS client. A host authenticated by this kind of
certificate can act as a client in a SIP over TLS connection when mutual authentication
is required by the server.
• TlsServer: The certificate identifies a TLS server. A host authenticated by this kind of
certificate can serve files or web pages using the HTTPS protocol or can act as a
server in a SIP over TLS connection.
 whether or not the certificate is owned by a CA (Certification Authority)
The Aastra unit uses two types of certificates:
Table 311: Certificates Types
Type Description
Host Certificates used to certify the unit (e.g.: a web server with HTTPS requires a host
certificate).
Others Any other certificate including trusted CA certificates used to certify peers (e.g.: a SIP
server with TLS).
The transferred certificate must be in Privacy Enhanced Mail (PEM) (host or others) or Distinguished Encoding
Rules (DER) (others) format. When transferring a host certificate, the certificate must be appended to the
private key to form one PEM file. The private key must not be encrypted.
You can transfer a certificate by using the HTTP or HTTPS protocol, but Aastra recommends to use HTTPS.
To access the unit via HTTPS, your browser must support RFC 2246 (TLS 1.0). The latest version of Microsoft
Internet Explorer supports HTTPS browsing.
Managing Certificates
You can view certificates information and you can delete certificates.

Chapter 43 - Certificates Management Certificate Authorities
 To view and manage certificates:

1. In the web interface, click the Management link, then the Certificates sub-link.
Figure 199: Management – Certificates Information Web Page
2
3
The Host Certificates section contains the certificates used to certify the unit. The Others
Certificates section contains any other certificate used to certify peers.
2. If applicable, delete a certificate in the Host Certificates or Others Certificates sections by clicking
the button of the certificate you want to delete.
3. If applicable, delete a certificate in the Other Certificates section by clicking the button of the
certificate you want to delete.
Certificate Authorities
This section contains information specific to certificate authority (CA) files.
 To view and manage certificate authorities information:

1. In the Certificate Authorities section of the Certificates page, define a specific OCSP URL to use for
certificate revocation status of certificates issued by this certificate authority (CA) in the
corresponding Override OCSP URL field.
Figure 200: Certificate Authorities Section
The URL should follow one of these formats:

http://hostname[:port]
http://hostname/path/filename.xxx
Note: The default empty value means that the OCSP URL present in the certificate to verify will be used
for checking its revocation status.
Certificate Upload through the Web Browser
The following steps explain how to transfer (add) a certificate from the web interface.

Certificate Upload through the Web Browser Software Configuration Guide
 To upload a certificate:
1. If you are currently using an unsecure HTTP access, the Certificate Upload Through Web Browser
section is disabled. This is to avoid transferring a certificate in clear text. To enable the section,
access the secure site by clicking the Activate unsecure certificate transfer through web browser
link at the top of the window.
2. In the Certificate Upload Through Web Browser section of the Certificates page, select the type of
the certificate in the Type drop-down menu.
Before transferring the certificate, you must indicate whether this is a Host or Others certificate.
Figure 201: Certificate Upload Through Web Browser Section
2 3 4
3. Use the Browse button to select the certificate to transfer.

The maximum certificate name is 50 characters.
4. Initiate the certificate upload by clicking the Upload Now button.
The Aastra unit immediately transfers the certificate. Once the certificate is transferred, you must
restart the SipEp and Web services in the System > Services page (“Chapter 4 - Services” on
page 53) before using the newly transferred certificate. Click the link in the message that is
displayed to access the Services web page.
Transferring a Certificate via Configuration Script

You can use a special command to transfer a certificate by configuration script or CLI. This command has the
following parameters: URL of the certificate to download, its Type (Host/ Others), the username and password.
See “Appendix B - Scripting Language” on page 627 for more details on the Aastra proprietary scripting
language
 To transfer a certificate via configuration script:

1. Use the following line in the CLI or a configuration script:
Cert.DownloadCertificate FileUrl=Value UserName=Value Password=Value Type=Value

Chapter 43 - Certificates Management Host Certificate Associations
where the different values may be as follows:

Table 312: Certificate Transfer Values
Value Description
FileUrl URL to a Certificate file that is loaded upon executing the execution of Download
command. The transfer protocols supported are:
• HTTP
• HTTPS
• TFTP
• FTP
Examples of valid URLS:
• http://www.myserver.com/Cert_MxDefault001.der
• tftp://myserver.com:69/myfolder/Cert_MxDefault001.der
When the port is not included in the URL, the default port for the chosen protocol is
used.
This field may contain some macros that are substituted by the actual value at the
moment of fetching the configuration script. The supported macros are:
• %mac% - the MAC address of the unit.
UserName When authentication is required by the remote file server, this variable is used as
the username.
Password When authentication is required by the remote file server, this variable is used as
the password.
Type Type of certificate to transfer.
• Host: Certificate used to certify the host system.
• Other: Remote systems certificates and issuers certificates.
For instance, a valid command would be:

Cert.DownloadCertificate FileUrl=http://www.myserver.com/Cert_MxDefault001.der
UserName=MyName Password=MyPassword Type=Host
Host Certificate Associations
The Host Certificate Associations section allows you to define which services can use the host certificates.
 To set host certificate associations:

1. In the Host Certificate Associations section of the Certificates page, check the services that can use
a given host certificate.
Figure 202: Host Certificate Associations Section
Table 313: Host Certificate Associations Parameters
SIP Specifies if this certificate can be used for SIP security.

Web Specifies if this certificate can be used for Web security.

Host Certificate Associations Software Configuration Guide
Table 313: Host Certificate Associations Parameters (Continued)
EAP Specifies if this certificate can be used for EAP security.

Chapter 43 - Certificates Management Host Certificate Associations

C H A P T E R
44 SNMP Configuration
This chapter describes how to set the SNMP parameters of the Aastra unit.
Standards Supported • RFC 1157: Simple Network Management Protocol (SNMP)

• RFC 1910: User-based Security Model for SNMPv2
• RFC 2104: HMAC: Keyed-Hashing for Message
Authentication
• RFC 2576: Coexistence between Version 1, Version 2, and
Version 3 of the Internet-standard Network Management
Framework
• RFC 2741: Agent Extensibility (AgentX) Protocol Version 1
• RFC 3411: An Architecture for Describing Simple Network
Management Protocol (SNMP) Management Frameworks
• RFC 3412: Message Processing and Dispatching for the
• RFC 3413: Simple Network Management Protocol (SNMP)
Applications
• RFC 3414: User-based Security Model (USM) for version 3 of
the Simple Network Management Protocol (SNMPv3)
• RFC 3415: View-based Access Control Model (VACM) for the
Simple Network Management Protocol (SNMP)a
• RFC 3416: Version 2 of the Protocol Operations for the
• RFC 3417: Transport Mappings for the Simple Network
Management Protocol (SNMP)
• RFC 3826: The Advanced Encryption Standard (AES) Cipher
Algorithm in the SNMP User-based Security Model
a. The Aastra unit complies to RFC 3415 but does not support the related MIBs.
Introduction
All parameters available in the Aastra unit web interface may also be configured via SNMP. The Aastra unit
SNMP feature offers the following options:
 Password-protected access
 Remote management
 Simultaneous management
The Aastra unit SNMP feature allows you to configure all the MIB services by using a SNMP browser to contact
the MIBs of the Aastra unit. It is assumed that you have basic knowledge of TCP/IP network administration.
Note: The Aastra unit’s SNMP settings do not support IPv6. See “IPv4 vs. IPv6 Availability” on page 85 for
more details.
You can use the MIB browser built in the Aastra’ Unit Manager Network.
You can also use any third-party SNMP browser or network management application running the SNMP
protocol to monitor and configure the Aastra unit. However, the information may not be presented in the same
manner depending on the SNMP browser used.

Chapter 44 - SNMP Configuration SNMP Configuration Section
Locate the proper parameter to modify and change (SET) its value.
SNMP Configuration Section
The SNMP Configuration section allows you to configure the SNMPv3 privacy information that allows securing
the Aastra unit, as well as defining where the Aastra unit must send traps.
 To set SNMP parameters:

1. In the web interface, click the Management link, then the Snmp sub-link.
Figure 203: Management – Snmp Web Page
4
5
6
7
2. Set the SNMP Listening Port field with the port number on which the SNMP service listens for
incoming SNMP requests.
The default value is 161.
3. Specify with which SNMP version a user can connect to the system by setting one of the following
drop-down menus to enable:
Table 314: SNMP Versions
SNMP Version Drop-down menu to set to Enable
SNMPv1 Enable SNMP V1

By default, SNMPv3 is enabled.
Caution: It is possible to disable all three versions of SNMP on the Aastra unit. If you do so, you will no
longer be able to access the unit in SNMP. To recover from this situation, you must perform a factory reset
procedure.
Note: Please note that a “public” user might be granted (unsecure) access by using SNMPv1 or SNMPv2,
while an “admin” user should rather be granted a SNMPv3 access. Furthermore, access for users in
SNMPv3 will require authentication and could be done with or without privacy according to the unit’s
configuration. This means that the unit does not grant an SNMPv3 access without authentication and
privacy.

SNMP Configuration Section Software Configuration Guide
4. If SNMPv3 is enabled, set the Authentication Protocol drop-down menu with the authentication
protocol to use with SNMPv3.
Table 315: SNMP Authentication Protocol
MD5 MD5 encoding is used. This is the default value.

SHA1 SHA1 encoding is used.
Caution: The Authentication Protocol field is not accessible if you have the User or Observer access right.
See “Users” on page 591 for more details.
SNMPv3 will grant access to all users who are configured in the unit and have a password with 8
characters or more (in the AAA service as described in “Chapter 49 - Access Control Configuration”
on page 591).
5. If SNMPv3 is enabled, set the privacy protocol to use with SNMPv3 in the Privacy Protocol drop-
down menu.
Table 316: SNMP Privacy Protocol
None No encryption is used. The Privacy Password parameter is ignored. This is the
default value.
DES DES encryption is used.
Caution: The Privacy Protocol field is not accessible if you have the User or Observer access right. See
“Users” on page 591 for more details.
6. If you are using the DES privacy, set the password to use in the Privacy Password field.
Caution: The Privacy Password field is not accessible if you have the User or Observer access right. See
7. Set the Community field with the string to use for the community field of SNMPv1 and SNMPv2
read-write commands and traps.
This field must not be empty.
The use of a community name provides context for agents receiving requests and initiating traps.
An SNMP agent won't respond to a request from a management system outside its configured
community.
The community name field may influence the AAA user name that will be used by the Aastra for
non-authenticated SNMP access (SNMPv1 and SNMPv2). See “Additional SNMP Parameters” on
page 441 for more information.
8. Specify that traps can be sent by setting the Enable SNMP Traps drop-down menu to enable.
There are five conditions that the Aastra unit checks before sending a trap:
• The traps are enabled.
• The destination address is valid.
• The NetSnmp Agent is ready.
• The destination address is reachable according to the routing table.
• The appropriate physical link is up.

Chapter 44 - SNMP Configuration SNMP Configuration Section
If all of those conditions are true, then the Aastra unit sends the traps. If any of those conditions is
false, the Aastra unit waits (1 second) and retries until it succeeds. Even if the traps are delayed,
they will be sent with the appropriate timestamp when all the conditions are met.
Furthermore, the SNMP version(s) currently enabled (see Step 2 for more details) define which type
of trap may be sent.
Table 317: Trap Type Sent vs SNMP Version Enabled
SNMP Version Enabled Trap Sent
SNMPv1 SNMPv2 SNMPv3 Trap V1 Trap V2c
Enabled
Enabled
Enabled Enabled
Enabled
Enabled Enabled
Enabled Enabled
Enabled Enabled Enabled
Note: You can also enable the traps via the CLI. See “Chapter 2 - Command Line Interface (CLI)” on
page 19 for details on how to work with the CLI.
The Aastra unit handles five different types of trap:

Table 318: Trap Types
Trap Description
coldStart A coldStart(0) trap means that the sending protocol entity is reinitializing
itself so that the agent's configuration or the protocol entity
implementation may be altered.
This trap is sent prior to a reboot that follows a firmware update, a
backup restoration or a default settings application. Note that if the unit is
shut down unexpectedly (power failure, power switch), this trap is not
emitted.
When the unit reboots because of a firmware upgrade, no coldStart traps
are sent before this reboot. In that specific case, a coldStart trap is sent
after the reboot if the installation scripts succeeded.
warmStart A warmStart(1) trap means that the sending protocol entity is reinitializing
itself so that neither the agent configuration nor the protocol entity
implementation is altered.
This trap is sent prior to all other reboots. Note that if the unit is shut
down unexpectedly (power failure, power switch), this trap is not emitted.
When the unit reboots because of a firmware upgrade, no warmStart
traps are sent before this reboot. In that specific case, a warmStart trap is
sent after the reboot if the installation scripts failed.
linkDown A linkDown(2) trap means that the SNMPv2 entity acting in an agent role
has detected that the ifOperStatus object for one of its communication
links is about to enter the down state from some other state. This other
state is indicated by the included value of ifOperStatus.
The Trap-PDU of type linkDown includes ifIndex, ifAdminStatus,
ifOperStatus (as of RFC 2233) of the interface that generated the trap.

SNMP Configuration Section Software Configuration Guide
Table 318: Trap Types (Continued)
Trap Description
linkUp A linkUp(3) trap means that the SNMPv2 entity, acting in an agent role,
has detected that the ifOperStatus object for one of its communication
links left the down state and transitioned into some other state (but not
into the notPresent state). This other state is indicated by the included
value of ifOperStatus.
The Trap-PDU of type linkUp includes ifIndex, ifAdminStatus,
ifOperStatus (as of RFC 2233) of the interface that generated the trap.
authenticationFailure An authenticationFailure(4) trap means that the sending protocol entity is
the addressee of a protocol message that is not properly authenticated.
This trap is sent when an authentication failure occurs from the Web, CLI
or SNMP interface.
9. If the traps are enabled, set the Trap Destination (s) field with the addresses/FQDNs and ports
where to send traps.
You can specify up to 5 destinations by using a comma between them (comma is not authorized
within a FQDN). The port numbers are optional. Note that the traps are sent simultaneously to all
destinations.
Example:
trapdest.com:2345, 123.45.67.89
Additional SNMP Parameters

A user name can be added to be used by the SNMP v1/v2 to access the configuration.
For non-authenticated access (SNMPv1 and SNMPv2), the Aastra will use the AAA user name from the
SnmpUser variable if it is not empty. If empty, the community name is used as the AAA user name.
 To add an SNMP user name:

1. In the snmpMIB, set the SnmpUser variable to a valid AAA user name.
snmp.SnmpUser="Value"
where Value is a valid AAA user name.
Caution: If the provided SNMP user name does not exist in the AAA.UsersStatus table or if the SNMP
user name is empty and the community name does not exist in the AAA.UsersStatus table, the SNMP
access will fail.
Partial Reset
When a partial reset is triggered, the following parameters are affected:
 Listening Port: Default value 161.
 Enable SNMPv1: Default value disable.
 Enable SNMPv2: Default value disable.

Chapter 44 - SNMP Configuration SNMP Statistics
 Enable SNMPv3: Default value enable.

See “Partial Reset” on page 15 for more details.
SNMP Statistics
The following are the statistics the Aastra unit keeps.

Table 319: SNMP Statistics
statsGetRequest Number of GET requests handled by the service.

statsGetNextRequest Number of GET-NEXT requests handled by the service.
statsSetRequest Number of SET requests handled by the service.

C H A P T E R
45 Access Control Configuration
This chapter describes how to set the Access Control parameters of the Aastra unit.
Standards Supported • RFC 2617: HTTP Authentication: Basic and Digest Access
Authentication
• RFC 2865: Remote Authentication Dial In User Service
(RADIUS)
• RFC 2866: RADIUS Accounting
Caution: The Access Control page is not accessible if you have the User or Observer access right. See
Users
The Users section allows you to manage the users that can access the web interface. You can add a maximum
of 10 users.
 To manage users:
1. In the web interface, click the Management link, then the Access Control sub-link.
Figure 204: Management – Access Control Web Page
2. If you want to add a new user, enter its name in the blank User Name field in the bottom left of the
window, enter the corresponding password in the blank Password field, then click the button.
The name is case-sensitive.
3. If you want to delete an existing user, click the corresponding button.
If you delete all users in the table, the profile’s default user(s) will be used upon unit restart.
Note: A system restart is required to completely remove the user. The current activities of this user are not
terminated on removal.
4. If you want to change the password of an existing user, type it in the corresponding Password field.
The password is case sensitive. All characters are allowed.
5. Define the access rights template applying to a user in the corresponding Access Rights drop-down
menu.

Chapter 45 - Access Control Configuration Services Access Control Type

Table 320: Access Rights
Access Right Description
Admin User is allowed to read and modify all variables of the unit.
User User is allowed to read and modify all variables except passwords and secrets.
Observer User is only allowed to read variables that are not passwords or secrets.
See “Access Rights Description” on page 447 for mode details on the various operations allowed
with each access right.
Partial Reset
When a partial reset is triggered, the password and access rights reset back to the default value (see “Partial
Reset” on page 15 for more details).
Services Access Control Type
The Services Access Control Type section allows you to define the type of authentication and accounting to
use for the CLI, SNMP, and Web services.
Authentication provides a way of identifying a user, typically by having the user enter a valid user name and
valid password before access is granted.
Accounting measures the resources a user consumes during access. This can include the amount of system
time or the amount of data a user has sent and/or received during a session.
 To set the Services Access Control type:

1. In the Services Access Control Type section of the Access Control page, set the authentication type
a service uses for incoming authentication requests in the corresponding Authentication Type
column.
Table 321: Authentication Types
Type Description
Local Incoming authentication attempts are validated against the user names and
passwords stored in the Local Users table (see “Users” on page 443 for more
details).
Radius Incoming authentication attempts are validated against the first responding
Radius server configured in the Radius Servers section (“Radius Servers” on
page 445). When no server is configured or the servers are unreachable, an
authentication attempt of type Local is performed against the user names and
passwords stored in the Local Users table (see “Users” on page 443 for more
details).
Note: This type is not available for the SNMP interface.
Figure 205: Access Control – Services Access Control Type Section

1 2

Radius Servers Software Configuration Guide
2. Set the accounting type a service uses in the corresponding Accounting Type column.
Accounting starts once users are successfully authenticated and stops when their session is over.
Table 322: Accounting Types
Type Description
None Accounting is disabled.

Radius Accounting is done by the first responding Radius server configured in the
Radius Servers section (“Radius Servers” on page 445).
Partial Reset
When a partial reset is triggered, the Radius authentication is disabled (see “Partial Reset” on page 15 for
more details).
Radius Servers
The Radius Servers section allows you to define up to three Radius servers. It also allows you to define
authentication server and accounting server information, for the CLI, SNMP, and Web services.
Note: The Aastra unit’s Radius server settings do not support IPv6. See “IPv4 vs. IPv6” on page 85 for more
details.
Radius Authentication occurs when the Authentication Type column of the Services Access Control Type
section (“Services Access Control Type” on page 444) is set to Radius for the service from which the
authentication request is coming. You can configure up to three Radius servers for each service listed in the
Select a Service drop-down menu. The first authentication attempt is sent to the Radius server with the highest
priority, which is set in the Priority column (1 being the highest priority). When authentication fails or the
request reaches the timeout set in the Server Request Timeout field, the next server with the highest priority
is used. When all servers have failed to reply or no servers are configured for the service asking for
authentication, authentication is attempted against local user names and passwords as a fallback strategy.
Radius authentication is available for the CLI and Web services.
Radius Accounting is enabled by setting the Accounting Type column of the Services Access Control Type
section (“Services Access Control Type” on page 444) to Radius for one or more services. When such a
configuration is set, accounting requests made through those services are forwarded to a Radius server
configured in the Radius Servers section. You can configure up to three Radius servers for each service listed
in the Select a Service drop-down menu. The first accounting request is sent to the Radius server with the
highest priority, which is set in the Priority column (1 being the highest priority). When the accounting request
fails or the request reaches the timeout set in the Server Request Timeout field, the next server with the
highest priority is used. The CLI, Web, and SNMP services can use the accounting functionality.
 To set the Radius servers information:

1. Select to which service you want to apply the changes in the Select a Service drop-down menu
above the Radius Servers section.
You can copy the configuration of the selected service to one or more services of the Aastra unit in
the Apply to the Following Services section at the bottom of the page. You can select specific
services by checking them, as well as use the Check All or Uncheck All buttons.
2. In the Authentication part of the Radius Servers section, set the host name and port of a Radius
server used for authentication requests in the corresponding Host field.

Chapter 45 - Access Control Configuration Radius Servers
Figure 206: Access Control – Radius Servers Section

1
2
3
4
5
6
7
You can configure up to three Radius servers with a different priority.
Note: This parameter is not available for the SNMP service.
3. Set the secret key shared between the Radius server and the unit in the corresponding Server
Secret field.
The Authentication Secret key must be the same as the secret key stored on the corresponding
Radius authentication server.
Note: This parameter is not available for the SNMP service.
4. In the Accounting part, set the host name and port of a Radius server used for accounting requests
in the corresponding Host field.
You can configure up to three Radius servers with a different priority.
5. Set the secret key shared between the Radius server and the unit in the corresponding Server
Secret field.
The Accounting Secret key must be the same as the secret key stored on the corresponding Radius
accounting server.
6. Set the Server Request Timeout field with the maximum time, in milliseconds, the unit waits for a
reply from a Radius server.
This parameter applies to all services. Upon reaching the timeout, the request is sent to the next
configured server.
7. Define the access rights template applying to a user in the corresponding Radius Users Access
Rights drop-down menu.
This parameter applies to all services. You have the following choices:
Table 323: Radius Users Access Rights
Access Right Description
Admin User is allowed to read and modify all variables of the unit.
User User is allowed to read and modify all variables except passwords and secrets.
Observer User is only allowed to read variables that are not passwords or secrets.
See “Access Rights Description” on page 447 for mode details on the various operations allowed
with each access right.

Access Rights Description Software Configuration Guide
Access Rights Description
You have three templates of rights from which you can select the permissions given to each user allowed in a
unit (see “Users” on page 443 and “Radius Servers” on page 445).
The following table describes the various operations allowed with each access right.
Table 324: Access Rights Description
Access Right Observer User Admin
Read configuration Yes Yes Yes

a
Modify Configuration No Yes Yes
Read/Write Passwords No No Yes
Change Access Rights No No Yes
Execute Configuration Script No Yesa Yes
Export Configuration No Yesa Yes
Backup/Restore No No Yes
configuration
Firmware updates No No Yes
a. Passwords cannot be changed and will not be exported to a configuration script.

Chapter 45 - Access Control Configuration Access Rights Description

C H A P T E R
46 File Manager
This chapter describes how to use the unit’s File Manager.
File Manager
The File page allows you to view and delete the files you have created with the File transfer protocol, for
instance, a configuration backup..
Note: The files under the File service are not included in the backup process. In the same way, the restore
process will not remove any file under the File service.
 To use the File manager:

1. In the web interface, click the Management link, then the File sub-link.
Figure 207: Management – File Web Page
If you want to delete an existing file, click the corresponding button.

You can directly download a file via your web browser by clicking it. You will then be able to see its
contents.
2. To add a file to the unit’s File System, type the path and name of the file to add in the field of the
Upload File Through Web Browser section, or select an existing one on the PC with the Browse
button.
If you are currently using an unsecure HTTP access, the Upload File Through Web Browser section
is disabled. This is to avoid transferring a file in clear text. To enable the section, access the secure
site by clicking the Activate unsecure file transfer through web browser link at the top of the window.
3. Click Submit.
Partial Reset
When a partial reset is triggered, the user-defined presets are deleted.

Chapter 46 - File Manager File Manager

C H A P T E R
47 Miscellaneous
This chapter describes how to set various parameters used to manage the Aastra unit.
Management Interface Configuration
The Miscellaneous page allows you to specify which one of the existing network interfaces is used to manage
the Aastra unit.
 To set the system management interface:

1. In the web interface, click the Management link, then the Misc sub-link.
Figure 208: Management – Misc Web Page
2. Select which one of the existing network interfaces is used to manage the device in the Network
Interface drop-down menu.
The management services (typically Web and/or SNMP) can be reached through this network
interface.
Before the system management services can be used, they need to be bound (or linked) to a
physical port of your Aastra unit.
The special value "All" means to bind all network interfaces.
Partial Reset
When a partial reset is triggered, the Management Interface reverts back to its default value.

Chapter 47 - Miscellaneous Management Interface Configuration

Appendices
A P P E N D I X
A Country-Specific Parameters
The following parameters differ depending on the country in which you are.
Definitions
The following are some useful definitions.
Table 325: Definitions
Term Description
Dial Tone Indicates the endpoint is ready to receive dialing.
Busy Tone Indicates the endpoint or equipment is in use, engaged or occupied.
Ringback Tone Indicates the called line is ringing out.
Special Information Tone Identifies network-provided announcements.
Stutter Dial Tone Notifies the user that they have a voice mail message when the phone
does not or cannot have a message-waiting light.
Confirmation Tone Confirms a command performed by the user (such as activate a

service).
Receiver Off Hook (ROH) Tone Indicates that the telephone is not hung up correctly.
Message Waiting Indicator Tone Indicates there is a message waiting somewhere for the owner of the
phone
Network Congestion Tone Indicates that all switching paths are busy, all toll trunks are busy, or
there are equipment blockages.
Intercept Tone Indicates that you have dialed incorrectly or that the feature you've
requested is not available on your terminal.
Preemption Tone In military telephone systems, a distinctive tone that is used to indicate
to connected users, i.e., subscribers, that their call has been
preempted by a call of higher precedence.
Reorder Tone Indicates that all switching paths are busy, all toll trunks are busy, there
are equipment blockages, the caller dialled an unassigned code, or the
digits dialled got messed up along the way.
FED Tone Indicates the far end tone detection.
Conventions
The following conventions apply to this Appendix.
Frequencies
 Symbol “*” means modulated. For instance: 425 Hz * 25 means 425 Hz modulated at 25 Hz.
 Symbol “+” means added. For instance: 425 Hz + 330 Hz means that both 425 Hz and 330 Hz

Appendix A - Country-Specific Parameters Definitions
sines are played at the same time.

 When a tone is composed of more than one frequency, if not otherwise specified, the given
electrical level applies to each frequency taken separately.
Impedance
Impedance is the apparent resistance, in an electric circuit, to the flow of an alternating current, analogous to
the actual electrical resistance to a direct current, being the ratio of electromotive force to the current.
When representing an impedance, the following applies:
 Symbol “//” means parallel.
 Symbol “+” means serial.
Furthermore, there are two types of impedances:
 Input Impedance
 Terminal Balance Return Loss (TBRL) Impedance
Input Impedance
Impedance of the Aastra at the Tip and Ring wires.
Terminal Balance Return Loss (TBRL) Impedance

Balance return loss attributable to transmission loss between two points. It is used to characterize an
impedance balancing property of the 2-wire analog equipment port.
Each country has its own definition of the TBRL value. For instance, in North America, TIA/EIA 464 (and TIA/
EIA 912) define two TBRL values:
 600 Ω for “on-premise” or short loop ports.
 350 Ω + (1000 Ω || 21 nF) for “off-premise” or long loop ports.
A wire length above 2.5 km is considered long loop according to TIA/EIA 912 section 6.4 (7)(b)).
In Europe, ETSI 300 439 also mentions a TBRL value. However, most European countries have different
requirements regarding the TBRL Impedance. This is also true for other countries around the world. Each one
of them has different requirements.
Line Attenuation
Values are given in dBr (deciBel relative):
 A “+” for input means that the digital side is attenuated by x decibels relative to the analog side.
 A “+” for output means that the analog side is amplified by x decibels relative to the digital side.
 A “-” for input means that the digital side is amplified by x decibels relative to the analog side.
 A “-” for output means that the analog side is attenuated by x decibels relative to the digital side.
On-Off Sequences
Values in bold are “on” cycles, where tones are audible. Values in normal style are “off” cycles, where tones
are not audible. When not otherwise specified, sequences repeat forever. A “x” symbol means that the
sequences between parenthesis is repeated x times. The next cycle(s) repeat forever, unless otherwise
specified. Values are in seconds.
For instance:
3*(0.1 – 0.1) then 0.6 – 1.0 - 0.2 – 0.2
means that the 0.1s on and 0.1s off sequence is repeated 3 times, afterwards the 0.6s on, 1.0s off, 0.2s on
and 0.2s off sequence repeats forever.

Australia Software Configuration Guide
Australia
The following parameters apply if you have selected Australia as location.
Australia 1
The following parameters apply if you have selected Australia 1 as location.
Table 326: Australia 1 Parameters
Parameter Value On – Off - CID Sequence (s) Elect. Levels
Busy Tone 425 Hz 0.38 – 0.38 -18 dBm
Call Waiting Tone 425 Hz 0.2 - 0.2, 0.2 - 4.4, 0.2 - 0.2, 0.2 - 4.4 -23 dBm
Dial Tone 425 Hz CONTINUOUS -18 dBm

400 Hz CONTINUOUS -24 dBm
Message Waiting Indicator Tone 425 Hz (0.1 - 0.04)x72, CONTINUOUS -18 dBm
400 Hz (0.1 - 0.04)x72, CONTINUOUS -24 dBm
Network Congestion Tone 425 Hz 0.38 - 0.38, 0.38 - 0.38 -13 dBm
425 Hz 0.38 - 0.38, 0.38 - 0.38 -23 dBm
Receiver Off Hook (ROH) Tone 2350 Hz CONTINUOUS -5 dBm
Reorder Tone 425 Hz 2.5 - 0.5 -18 dBm
Ringback Tone 425 Hz 0.4 - 0.2, 0.4 - 2.0 -18 dBm

400 Hz 0.4 - 0.2, 0.4 - 2.0 -24 dBm
450 Hz 0.4 - 0.2, 0.4 - 2.0 -24 dBm
Special Information Tone 425 Hz 2.5 - 0.5 -18 dBm
Stutter Dial Tone 425 Hz CONTINUOUS -18 dBm

Ring AC: 53 VRMS, 25 Hz 0.4 - 0.2, 0.4 - 2.0

DC: -10 Vdc
Loop Current 30 ma
Input Impedance (FXS) 600 Ω
Input Impedance (FXO) 600 Ω
Tbrl-Impedance 600 Ω
FED Tone 425 Hz 8.0
Default Caller ID (FXS) BELLCORE
FXS Line Attenuation (Input) +0 dBr
FXS Line Attenuation (Output) -6 dBr
FXO Line Attenuation (Input) +6 dBr
FXO Line Attenuation (Output) +0 dBr
Delay Before Answering 2 seconds
Delay Before Dialing (No Dial 4 seconds

Tone Detection)

Appendix A - Country-Specific Parameters Australia
Australia 2
The following parameters apply if you have selected Australia 2 as location.
Table 327: Australia 2 Parameters
Busy Tone 425 Hz 0.38 – 0.38 -18 dBm
Call Waiting Tone 425 Hz 0.2 - 0.2, 0.2 - 4.4, 0.2 - 0.2, 0.2 - 4.4 -23 dBm

Network Congestion Tone 425 Hz 0.38 - 0.38, 0.38 - 0.38 -13 dBm
425 Hz 0.38 - 0.38, 0.38 - 0.38 -23 dBm
Receiver Off Hook (ROH) Tone 2350 Hz CONTINUOUS -5 dBm
Reorder Tone 425 Hz 2.5 - 0.5 -18 dBm
Ringback Tone 425 Hz 0.4 - 0.2, 0.4 - 2.0 -18 dBm

400 Hz 0.4 - 0.2, 0.4 - 2.0 -24 dBm
450 Hz 0.4 - 0.2, 0.4 - 2.0 -24 dBm
Special Information Tone 425 Hz 2.5 - 0.5 -18 dBm
Stutter Dial Tone 425 Hz CONTINUOUS -18 dBm

Ring AC: 53 VRMS, 25 Hz 0.4 - 0.2, 0.4 - 2.0

DC: -10 Vdc
Loop Current 30 ma
Input Impedance (FXS) 220 Ω + 820 Ω // 120 nF
Input Impedance (FXO) 220 Ω + 820 Ω // 115 nF
Tbrl-Impedance 220 Ω + 820 Ω // 120 nF
FED Tone 425 Hz 8.0
FXS Line Attenuation (Input) -3 dBr

Tone Detection)

Austria Software Configuration Guide
Austria
The following parameters apply if you have selected Austria1 as location.

Table 328: Austria1 Parameters
Busy Tone 450 Hz 0.3 – 0.3 -20 dBm
Call Waiting Tone 440 Hz 2.0 – 0.3, 10.0 – 0.3, 10.0 -17 dBm
Confirmation Tone 450 Hz (0.1 – 0.1) x 3 End -20 dBm
Message Waiting Indicator Tone 450 Hz (0.1 – 0.1) x 10, CONTINUOUS -20 dBm
Network Congestion Tone 450 Hz 0.3 – 0.3 -20 dBm
Receiver Off Hook (ROH) Tone 1400+2060+2450+2600 Hz 0.1 – 0.1 -19 dBm
Reorder Tone 950 Hz 0.33 - 0.33 - 0.33 - 1.0 -20 dBm

1400 Hz 0.33 - 0.33 - 0.33 - 1.0 -20 dBm
1800 Hz 0.33 - 0.33 - 0.33 - 1.0 -20 dBm
Ringback Tone 450 Hz 1.0 – 5.0 -20 dBm
Special Information Tone 950 Hz 0.33 - 0.33 - 0.33 - 1.0 -20 dBm
1400 Hz 0.33 - 0.33 - 0.33 - 1.0 -20 dBm
1800 Hz 0.33 - 0.33 - 0.33 - 1.0 -20 dBm
Stutter Dial Tone 450 Hz (0.1 – 0.1) x 3, CONTINUOUS -20 dBm
Ring (FXS) AC: 45 VRMS, 50 Hz 1.0 – 5.0

DC: -15 Vdc
Loop Current 30 ma
Flash Hook Detection Range Min: 170 ms

Max: 900 ms
FED Tone 450 Hz 8.0
FXO Line Attenuation (Output) -1 dBr

Tone Detection)

Appendix A - Country-Specific Parameters Brazil
Brazil
The following parameters apply if you have selected Brazil as location.

Table 329: Brazil Parameters
Busy Tone 425 Hz 0.25 – 0.25 -10 dBm
Receiver Off Hook (ROH) Tone 425 Hz 0.25 – 0.25 -10 dBm
Reorder Tone 425 Hz 0.75 – 0.25, 0.25 – 0.25 -10 dBm
Special Information Tone 950 Hz 0.33 - 0.03 - 0.33 - 0.03 - 0.33 - 1.0 -15 dBm
1400 Hz 0.33 - 0.03 - 0.33 - 0.03 - 0.33 - 1.0 -15 dBm
1800 Hz 0.33 - 0.03 - 0.33 - 0.03 - 0.33 - 1.0 -15 dBm

DC: -15 Vdc
Loop Current 30 ma

Max: 900 ms
Tbrl-Impedance 800 Ω // 50 nF
FED Tone 425 Hz 8.0
Default Caller ID (FXS) TELEBRAS_DTMF
FXS Line Attenuation (Input) 0 dBr
FXO Line Attenuation (Input) 6 dBr
FXO Line Attenuation (Output) 0 dBr

Tone Detection)

China Software Configuration Guide
China
The following parameters apply if you have selected China as location.

Table 330: China Parameters
Busy Tone 450 Hz 0.35 – 0.35 -10 dBm
Call Waiting Tone 450 Hz 0.4 – 4.0, 0.4 – 4.0 -20 dBm
Confirmation Tone 450 Hz (0.1 – 0.1) x 3, End -10 dBm
Intercept Tone 450 Hz 0.2 – 0.2, 0.2 – 0.6 -20 dBm
Message Waiting Indicator Tone 450 Hz 0.4 – 0.04 -10 dBm
Preemption Tone 450 Hz 0.2 – 0.2, 0.2 – 0.6 -20 dBm
Receiver Off Hook (ROH) Tone 950 Hz 5.0 – 5.0 – 5.0 – 5.0 -25 dBm
950 Hz 5.0 – 5.0 – 5.0 – 5.0 -16 dBm
950 Hz 5.0 – 5.0 – 5.0 – 5.0 -8 dBm
950 Hz 5.0 – 5.0 – 5.0 – 5.0 -6 dBm
Reorder Tone 450 Hz 0.1 – 0.1, 0.1 – 0.1, 0.1 – 0.1, 0.4 – 0.4 -10 dBm
1400 Hz 0.33 - 0.33 - 0.33 - 1.0 -10 dBm
1800 Hz 0.33 - 0.33 - 0.33 - 1.0 -10 dBm
Stutter Dial Tone 450 Hz 0.4 – 0.04 -10 dBm
Ring AC: 45 VRMS, 25 Hz 1.0 – 4.0

DC: -15 Vdc
Loop Current 30 ma

Max: 900 ms
FED Tone 450 Hz 8.0
Default Caller ID BELLCORE

Tone Detection)

Appendix A - Country-Specific Parameters Czech Republic
Czech Republic
The following parameters apply if you have selected Czech Republic1 as location.
Table 331: Czech Republic1 Parameters
Busy Tone 425 Hz 0.33 – 0.33 -12 dBm
Dial Tone 425 Hz 0.33 – 0.33, 0.66 – 0.66 -12 dBm
Message Waiting Indicator Tone 425 Hz (0.1 – 0.1) x 10, 0.33 – 0.33, 0.66 – 0.66 -12 dBm
Receiver Off Hook (ROH) Tone 425 Hz 0.17 – 0.17 -12 dBm
1400 Hz 0.33 - 0.33 - 0.33 - 1.0 -12 dBm
1800 Hz 0.33 - 0.33 - 0.33 - 1.0 -12 dBm
Stutter Dial Tone 425 Hz (0.17 – 0.17) x 3, 0.66 – 0.66 -12 dBm

DC: -15 Vdc
Loop Current 30 ma

Max: 900 ms
Tbrl Impedance 220 Ω + 820 Ω // 115 nF
FED Tone 425 Hz 0.165 – 0.165
Default Caller ID (FXS) V23

Tone Detection)

France Software Configuration Guide
France
The following parameters apply if you have selected France1 as location.

Table 332: France1 Parameters
Busy Tone 440 Hz 0.5 – 0.5 -20 dBm
Special Information Tone 950 Hz 0.3 - 0.03 - 0.3 - 0.03 - 0.3 - 1.0 -20 dBm
1400 Hz 0.3 - 0.03 - 0.3 - 0.03 - 0.3 - 1.0 -20 dBm
1800 Hz 0.3 - 0.03 - 0.3 - 0.03 - 0.3 - 1.0 -20 dBm

DC: -15 Vdc
Loop Current 30 ma

Max: 900 ms
FED Tone 440 Hz 8.0
FXS Line Attenuation (Input) +1.9 dBr
FXS Line Attenuation (Output) -8.9 dBr

Tone Detection)

Appendix A - Country-Specific Parameters Germany
Germany
The following parameters apply if you have selected Germany as location.
Germany 1
The following parameters apply if you have selected Germany 1 as location.
Table 333: Germany 1 Parametersa
Busy Tone 425 Hz 0.48 – 0.48 -16 dBm
Call Waiting Tone 440 Hz 0.3 – End -17 dBm
1400 Hz 0.33 - 0.33 - 0.33 - 1.0 -16 dBm
1800 Hz 0.33 - 0.33 - 0.33 - 1.0 -16 dBm

DC: -15 Vdc
Loop Current 30 ma

Max: 900 ms
FED Tone 425 Hz 8.0

Tone Detection)
a. The Germany 2 choice in the MIB is exactly the same as Germany 1.

Germany Software Configuration Guide
Germany 2
The following parameters apply if you have selected Germany 2 as location.
Table 334: Germany 2 Parameters
Busy Tone 425 Hz 0.48 – 0.48 -13 dBm
1400 Hz 0.33 - 0.33 - 0.33 - 1.0 -13 dBm
1800 Hz 0.33 - 0.33 - 0.33 - 1.0 -13 dBm

DC: -5 Vdc
Loop Current 30 ma

Max: 900 ms
FED Tone 425 Hz 8.0

Tone Detection)

Appendix A - Country-Specific Parameters Israel2
Israel2
The following parameters apply if you have selected Israel2 as location.

Table 335: Israel2 Parameters
Busy Tone 400 Hz 0.5 – 0.5 -14 dBm
Call Waiting Tone 400 Hz 0.5 – 10.0, 0.5 – 10.0 -16 dBm
Confirmation Tone 400 Hz 0.17 – 0.34, 0.14 – 0.14, End -14 dBm
Hold Tone 400 Hz 0.05 – 2.0, End -16 dBm
Reorder Tone 1000 Hz 0.33 - 0.33 - 0.33 - 1.0 -14 dBm

1400 Hz 0.33 - 0.33 - 0.33 - 1.0 -14 dBm
1800 Hz 0.33 - 0.33 - 0.33 - 1.0 -14 dBm
Special Information Tone 450 + 150 Hz 0.5 – End -14 dBm
Ring AC: 45 VRMS, 25 Hz 1.0 – 3.0

DC: -15 Vdc
Loop Current 30 ma

Max: 900 ms
Tbrl-impedance 600 Ω
FED Tone 400 Hz 8.0

Tone Detection)

Italy Software Configuration Guide
Italy
The following parameters apply if you have selected Italy1 as location.

Table 336: Italy1 Parameters
Busy Tone 425 Hz 0.5 – 0.5 -13 dBm
Dial Tone 425 Hz 0.2 – 0.2, 0.6 – 1.0 -13 dBm
Message Waiting Indicator Tone 425 Hz (0.1 – 0.1) x 10, 0.2 – 0.2, 0.6 – 1.0 -13 dBm
1400 Hz 0.33 - 0.33 - 0.33 - 1.0 -20 dBm
1800 Hz 0.33 - 0.33 - 0.33 - 1.0 -20 dBm
Stutter Dial Tone 425 Hz (0.1 – 0.1) x 3, 0.2 – 0.2, 0.6 – 1.0 -13 dBm

DC: -15 Vdc
Loop Current 30 ma

Max: 900 ms
Tbrl-impedance 750 Ω // 18 nF
FED Tone 425 Hz 0.2 – 0.2, 0.6 – 1.0

Tone Detection)

Appendix A - Country-Specific Parameters Japan
Japan
The following parameters apply if you have selected Japan 2 as location.

Table 337: Japan 2 Parameters
Busy Tone 400 Hz 0.5 – 0.5 -13 dBm
Call Waiting Tone 400 Hz 2.0 - 0.3, 10.0 - 0.3, 10.0 -17 dBm

420 Hz 1.0 – 2.0 -22 dBm
380 Hz 1.0 – 2.0 -22 dBm
Special Information Tone 400 Hz 0.1 – 0.1 -13 dBm
Stutter Dial Tone 400 Hz (0.1 - 0.1)x3, CONTINUOUS -13 dBm
Ring AC: 45 VRMS, 20 Hz 1.0 – 2.0

DC: -15 Vdc
Loop Current 30 ma
Input Impedance (FXS) 600 Ω + 1000 nF
Tbrl-Impedance 600 Ω + 1000 nF
FED Tone 400 Hz 8.0
FXS Line Attenuation (Input) +0 dBr

Tone Detection)

Mexico Software Configuration Guide
Mexico
The following parameters apply if you have selected Mexico as location.

Table 338: Mexico Parameters
Busy Tone 425 Hz 0.25 – 0.25 -18 dBm
Message Waiting Indicator Tone 425 Hz (0.1 – 0.1) x 10 CONTINUOUS -14 dBm
Preemption Tone 425 Hz 0.5 – 0.17, 0.17 – 0.17 -18 dBm
1400 Hz 1.0 - 1.0 - 1.0 - 1.0 -14 dBm
1800 Hz 1.0 - 1.0 - 1.0 - 1.0 -14 dBm
Ring AC: 45 VRMS, 25 Hz 1.0 – 4.0

DC: -15 Vdc
Loop Current 30 ma

Max: 900 ms
FED Tone 425 Hz 8.0
Delay Before Answering 0 second
Delay Before Dialing (No Dial 0 second

Tone Detection)

Appendix A - Country-Specific Parameters North America
North America
The following parameters apply if you have selected North America as location.
North America 1
The following parameters apply if you have selected North America 1 as location.
Table 339: North America 1 Parameters
Busy Tone 480+620 Hz 0.5 – 0.5 -21 dBm
Confirmation Tone 350+440 Hz (0.1 – 0.1) x 3, End -17 dBm
Dial Tone 350+440 Hz CONTINUOUS -17 dBm
Intercept Tone 440+620 Hz 0.5 – 0.5 -14 dBm
Message Waiting Indicator Tone 350+440 Hz (0.1 – 0.1) x 10, CONTINUOUS -17 dBm
Network Congestion Tone 480+620 Hz 0.25 – 0.25 -21 dBm
Reorder Tone 480+620 Hz 0.3 – 0.2 -21 dBm
Ringback Tone 440+480 Hz 2.0 – 4.0 -19 dBm
1400 Hz 0.33 - 0.33 - 0.33 - 1.0
1800 Hz 0.33 - 0.33 - 0.33 - 1.0
Stutter Dial Tone 350+440 Hz (0.1 – 0.1) x 3, CONTINUOUS -17 dBm

DC: -15 Vdc
Loop Current 30 ma

Max: 1100 ms
Tbrl-Impedancea (FXS) 600 Ω
FED Tone 440 Hz 8.0
Delay Before Dialing (No Dial 0.7 seconds

Tone Detection)
a. TBRL-Impedance for “on-premise” or short loop ports.

Spain Software Configuration Guide
Spain
The following parameters apply if you have selected Spain1 as location.

Table 340: Spain1 Parameters
Busy Tone 425 Hz 0.2 – 0.2 -13 dBm
Network Congestion Tone 425 Hz 0.2 – 0.2, 0.2 – 0.2, 0.2 – 0.6 -13 dBm
Reorder Tone 425 Hz 0.2 – 0.2, 0.2 – 0.6 -13 dBm
1400 Hz 0.33 - 0.33 - 0.33 - 1.0 -20 dBm
1800 Hz 0.33 - 0.33 - 0.33 - 1.0 -20 dBm

DC: -15 Vdc
Loop Current 30 ma

Max: 900 ms
FED Tone 425 Hz 8.0

Tone Detection)

Appendix A - Country-Specific Parameters Switzerland
Switzerland
The following parameters apply if you have selected Switzerland as location.

Table 341: Switzerland Parameters
Busy Tone 425 Hz 0.5 – 0.5 -13 dBm
1400 Hz 0.33 - 0.33 - 0.33 - 1.0 -13 dBm
1800 Hz 0.33 - 0.33 - 0.33 - 1.0 -13 dBm

DC: -15 Vdc
Loop Current 30 ma

Max: 900 ms
Tbrl-impedance 220 Ω + 820 Ω // 115 nF
FED Tone 425 Hz 8.0
FXS Line Attenuation (Output) -6.5 dBr

Tone Detection)

United Arab Emirates Software Configuration Guide
United Arab Emirates
The following parameters apply if you have selected United Arab Emirates as location.
United Arab Emirates 2

The following parameters apply if you have selected the United Arab Emirates 2 as location.
Table 342: United Arab Emirates 2 Parameters
Busy Tone 400 Hz 0.38 – 0.38 -13 dBm
Call Waiting Tone 425 Hz (0.2 – 12.0, 0.2 –12.0)x2 End -13 dBm
Message Waiting Indicator Tone 350+440 Hz (0.1 – 0.1) x 10 CONTINUOUS -13 dBm
Network Congestion Tone 400 Hz 0.4 – 0.35, 0.23 – 0.53 -13 dBm
Receiver Off Hook (ROH) Tone 1400+2060+2450+2600 Hz (0.1 – 0.1) -19 dBm
Reorder Tone 400 Hz CONTINUOUS -13 dBm
Ringback Tone 425 Hz 0.4 – 0.2, 0.4 – 2.0 -13 dBm
Special Information Tone 950 Hz 0.33 – 0.33, 0.33 – 1.0 -13 dBm
1400 Hz 0.33 – 0.33, 0.33 – 1.0 -13 dBm
1800 Hz 0.33 – 0.33, 0.33 – 1.0 -13 dBm
Stutter Dial Tone 350+450 Hz (0.4 – 0.04-) x 5 CONTINUOUS -13 dBm
Ring AC: 45 VRMS, 25 Hz 0.4 – 0.2, 0.4 – 2.0

DC: -15 Vdc
Loop Current 30 ma

Max: 900 ms
FED Tone 440 Hz 8.0

Tone Detection)

Busy Tone 400 Hz 0.38 – 0.38 -19 dBm

Appendix A - Country-Specific Parameters United Arab Emirates
Table 343: United Arab Emirates 3 Parameters (Continued)
Call Waiting Tone 440 Hz 2.0 – 0.3, 10.0 –0.3, 10.0 -17 dBm
Confirmation Tone 350+440 Hz (0.1 – 0.1) x 3 End -22 dBm
Ringback Tone 400+450 Hz 0.4 – 2.0, 0.4 – 0.2 -22 dBm
1400 Hz 0.33 – 0.33, 0.33 – 1.0 -19 dBm
1800 Hz 0.33 – 0.33, 0.33 – 1.0 -19 dBm
Ring AC: 45 VRMS, 25 Hz 0.4 – 2.0, 0.4 – 0.2

DC: -15 Vdc
Loop Current 30 ma

Max: 900 ms
FED Tone 440 Hz 8.0

Tone Detection)

Busy Tone 400 Hz 0.38 – 0.38 -19 dBm
Call Waiting Tone 440 Hz 2.0 – 0.3, 10.0 –0.3, 10.0 -17 dBm
Confirmation Tone 350+440 Hz (0.1 – 0.1) x 3 End -22 dBm

United Arab Emirates Software Configuration Guide
Table 344: United Arab Emirates 4 Parameters (Continued)
1400 Hz 0.33 – 0.33, 0.33 – 1.0 -19 dBm
1800 Hz 0.33 – 0.33, 0.33 – 1.0 -19 dBm
Ring AC: 45 VRMS, 25 Hz 0.4 – 2.0, 0.4 – 0.2

DC: -15 Vdc
Loop Current 30 ma

Max: 900 ms
FED Tone 440 Hz 8.0

Tone Detection)

Appendix A - Country-Specific Parameters UK
UK
The following parameters apply if you have selected the United Kingdom as location.
Table 345: UK Parameters
Busy Tone 400 Hz 0.38 – 0.38 -19 dBm
Confirmation Tone 350+440 Hz (0.1 – 0.1) x 3, End -22 dBm
Message Waiting Indicator Tone 350+440 Hz (0.1 – 0.1) x 10, CONTINUOUS -22 dBm
1400 Hz 0.33 – 0.33, 0.33 – 1.0 -19 dBm
1800 Hz 0.33 – 0.33, 0.33 – 1.0 -19 dBm
Stutter Dial Tone 350+440 Hz (0.1 – 0.1) x 3, CONTINUOUS -22 dBm
Ring (FXS) AC: 45 VRMS, 25 Hz 0.4 – 0.2, 0.4 – 2.0

DC: -15 Vdc
Loop Current 30 ma

Max: 900 ms
Tbrl-impedance 370 Ω + 620 Ω // 310 nF
FED Tone 440 Hz 8.0
Default Caller ID (FXS) V23

Tone Detection)

A P P E N D I X
B Scripting Language
This appendix describes the Aastra proprietary scripting language. It also lists a few configuration samples
that can be pasted or typed into the CLI (see “Chapter 2 - Command Line Interface (CLI)” on page 11 for more
details) or downloaded into the Aastra via the Configuration Script feature (see“Chapter 40 - Creating a
Configuration Script” on page 414.
You can substitute the values listed in these examples with your own values. When enums are involved, refer
to the MIB structure with a MIB browser to determine the actual value you need to insert. You can also refer
to the Configuration Reference Guide, which lists all the parameters, tables, and commands available in the
Aastra.
This appendix covers the following topics:
 General Scripting Language Syntax
 Assigning scalar values
 Assigning table cell values
 Executing commands
 Variable Values (Enums)
 Call Router Specific Information
 Examples
General Scripting Language Syntax
The Aastra proprietary scripting language can be used to assign values to configuration variables and execute
configuration commands. The scripting language may be used when creating configuration scripts and when
working with the Command Line Interface (“Chapter 2 - Command Line Interface (CLI)” on page 11).
Using the scripting language requires a bit of knowledge about the Aastra’s configuration variables tree
structure.
The scripting language uses the following general syntax:
[keyword] [Context_Name [separator expression [operator constant]]] [#comment]
All specific syntaxes in this Appendix are derived from this general syntax.
Note that the brackets ([ and ]) are used to mark optional arguments. They are not part of the syntax.
Table 346: Scripting Language Syntax
Token Types Description
keyword A token that defines the type of operation to execute on expression or the type of data to
retrieve from expression. Currently, only the set keyword is supported, which assigns
value of constant to expression.
Context_Name Defines to which service the following expression belongs. For instance, the
Configuration Manager context is Conf, the Firmware Pack Updater context is Fpu, and
the Host Configuration context is Hoc.
separator Delimiter defined as “.” (dot).

Appendix B - Scripting Language General Scripting Language Syntax
Table 346: Scripting Language Syntax (Continued)
Token Types Description
expression String that describes a configuration object. It can resolve to either:

• a scalar variable
• a cell
• a column
• a row
• a table
When a service, a scalar, a command or a table variable has the same name as a
keyword, you should use the "get" and "set " keywords to access this variable.
operator Only the assignment operator (=) is defined.
constant A textual string or a number to assign to expression.
comment Anything following the comment marker (#) up to the end of the line is ignored.
Supported Characters
When using the scripting language, the following ASCII codes are supported:
10 LF, line feed 62 >, greater than 94 ^, caret
13 CR, carriage return 63 ?, question mark 95 _, underscore
32 space 64 @, commercial at 96 `, back quote
33 !, exclamation mark 65 A 97 a
34 ", double quote 66 B 98 b
35 #, hash 67 C 99 c
36 $, dollar 68 D 100 d
37 %, percent 69 E 101 e
38 &, ampersand 70 F 102 f
39 ', quote 71 G 103 g
40 (, open parenthesis 72 H 104 h
41 ), close parenthesis 73 I 105 i
42 *, asterisk 74 J 106 j
43 +, plus 75 K 107 k
44 ,, comma 76 L 108 l
45 -, minus 77 M 109 m
46 ., full stop 78 N 110 n
47 /, oblique stroke 79 O 111 o
48 0, zero 80 P 112 p
49 1 81 Q 113 q
50 2 82 R 114 r
51 3 83 S 115 s
52 4 84 T 116 t
53 5 85 U 117 u
54 6 86 V 118 v
55 7 87 W 119 w
56 8 88 X 120 x
57 9 89 Y 121 y
58 :, colon 90 Z 122 z
59 ;, semicolon 91 [, open square bracket 123 {, open curly bracket
60 <, less than 92 \, backslash 124 |, vertical bar
61 =, equals 93 ], close square bracket 125 }, close curly bracket
126 ~, tilde
All other ASCII codes are invalid.

Note that you must escape the XML reserved characters when inserting them in a configuration script:
 < : <
 > : >
 % : %

Assigning Scalar Values Software Configuration Guide
 & : &
Assigning Scalar Values
The following is a sample script command assigning a value to a scalar configuration variable:
Service_Name.Scalar_Name=value
Table 347: Scalar Syntax
Command Description
Service_Name Defines which service should process the expression. For instance, the
Configuration Manager context is Conf, the Firmware Pack Updater context is
Fpu, and the Host Configuration context is Hoc.
Scalar_Name Name of the specific variable to which assign a value.
value A textual string or a number to assign to the argument.
A valid script line would be:

Cli.InactivityTimeOut=25
Assigning Table Cell Values
When you want to get the value of a specific table cell or set the value of a specific table cell, you must follow
a particular syntax:
[get]Service_Name.Table_Name[Index=key].Column_Name
[set]Service_Name.Table_Name[Index=key].Column_Name=<value>
Table 348: Table Cell Syntax
Command Description
Service_Name Defines which service should process the expression. For instance, the
Table_Name Name of the table that contains the cell.
index=key List of index values identifying the row on which the cell is located. The list of
indexes is in the name=value form, separated by spaces and enclosed within
brackets. The index is always the first column of a table.
Column_Name Name of the column that contains the cell.
value A textual string or a number to assign to the argument.
Let’s take for instance the NetworkInterfacesStatus table:
InterfaceName InterfaceStatus LinkName IpAddr
Interface1 100a 10.1.1.1
Interface2 400b 10.1.1.2
Interface3 400 cd
10.1.1.3
a. This enum means “disabled”

Appendix B - Scripting Language Executing Commands
b. This enum means “ok”

c.
d.
If you want to get the IP address value of Interface 3, you would have to enter the following command:
get Bni.NetworkInterfacesStatus[InterfaceName=Interface3].IpAddr
Executing Commands
Configuration commands are used to make the Aastra perform actions such as restarting the unit, restarting
a service, refreshing its SIP registration, etc.
There are two types of commands you can execute:
 Normal Commands
 Row Commands
Normal Commands
The normal command feature has the following syntax:
Service_Name.Command_Name arg1=value1 –b arg2=[value2 value3 value4]
Table 349: Normal Command Syntax
Command Description
Service_Name Defines which service should process the command. For instance, the
Command_Name The command to execute.
argn Name of the argument for which you want to assign a value. Three types of
arguments are allowed:
• flags (beginning with ‘-‘, without anything else, for instance, “-b” in the
command syntax above). Flags are optional.
• scalar arguments (with mandatory ‘=’ and following value). They are
mandatory unless they have a default value, in which case they are
optional.
• vector arguments (with mandatory ‘=’ and following a list of values
enclosed within brackets and separated by spaces). They are
mandatory unless they have a default value, in which case they are
optional.
The number and types of arguments depend on the specific command you are
using.
valuen A textual string or a number to assign to the argument.
For instance, a valid command would be:

Conf.BackupImage FileName=backup_test_1 Location=/testfiles/Conf/v1/ManualTests/
TransferProtocol=400 TransferUsername=testuser TransferPassword=test
TransferSrvHostname=”test1.Aastra.com”
Another valid command (without arguments) would be:
SipEp.RegistrationRefresh
Double Quotes

Variable Values (Enums) Software Configuration Guide
You must use double quotes when the text parameter contains special characters such as dot or "#". For
instance, entering the following command results in a bad command:
Conf.BackupImage FileName=test.cfg Location=config TransferProtocol=400
TransferUsername=Usr1 TransferPassword=Pwd1 TransferSrvHostname=192.168.6.3
You must enclose each text parameter that contains special characters such as dot or "#" with double quotes.
In the above example, you must enclose FileName=test.cfg and TransferSrvHostname=192.168.6.3 in double
quotes:
Conf.BackupImage FileName="test.cfg" Location=config TransferProtocol=400
TransferUsername=Usr1 TransferPassword=Pwd1 TransferSrvHostname="192.168.6.3"
Row Commands
Row commands appear as table cells and allow you to perform an action on a specific row of the relevant table.
Row commands are available in several services of the Aastra. For instance, the Call Router service uses the
Up, Down, Insert, and Delete commands in its various tables.
The row command feature has the following syntax:
Context_Name.Table_Name[index1=value1 index2=value2].Row_Command=execute_value
Table 350: Row Command Syntax
Command Description
Context_Name Defines which service should process the command. For instance, the
Table_Name Table where the row command is located.
indexn=valuen List of index values identifying the row on which to execute the command. The
list of indexes is in the name=value form, separated by spaces and enclosed
within brackets. The index is always the first column of a table. See “Assigning
Table Cell Values” on page 479 for more details.
Row_Command The row command to execute.
execute_value Numerical value of the enum.
For instance, the following executes the service Dhcp’s StaticLeases Delete row command on one of the
table’s rows. The StaticLeases table only has one index column: the MacAddress column. The Delete row
command is an enum that has two possible values: noOp (0) and delete (10). The command is executed by
assigning the execute value (10) to the Delete cell.
Dhcp.StaticLeases[MacAddress="0090F8001234"].Delete=10
DeleteAllRows Command
The DeleteAllRows command is a table command that you can use to delete all rows of a specific table to start
anew. You can use it as follows:
Service_Name.Table_Name.DeleteAllRows
A valid command would be:
CRout.MappingExpression.DeleteAllRows
Variable Values (Enums)
The scripting language represents enums with their numeric value, and not their textual value. For instance,
the TFTP transfer protocol values available are as follows:
 100
 200

Appendix B - Scripting Language Call Router Specific Information
 300
 400
 500
This does not mean much. By looking into the MIB structure of the Aastra with a MIB browser or requesting
help on the variable in the CLI, you will be able to determine that the values really mean the following:
 100: HTTP
 200: HTTPS
 300: TFTP
 400: FTP
 500: FILE
Call Router Specific Information
When working with call router parameters, you must be aware of the following:
 You must prefix the name of a route with “route-”, for instance: route-isdn_sip.
 You must prefix the name of a SIP interface with “sip-”, for instance: sip-default.
 You must prefix the name of an ISDN interface with “isdn-”, for instance: isdn-default.
 You must prefix the name of a hunt with “hunt-”, for instance: hunt-hunt1.
Examples
This section gives a few configuration samples that can be used both in the CLI or as part of a configuration
script.
Management Functions
The following sections describe how to perform some useful management functions such as a configuration
backup/restore and changing the default user password.
Configuration Backup / Restore

Each of the two following commands must be created in one line.
Conf.BackupImage FileName="image.text" Location="resultfolder" TransferProtocol=300
TransferUsername="" TransferPassword="" TransferSrvHostname=“192.168.3.4"
Conf.RestoreImage FileName="image.text" Location="resultfolder" TransferProtocol=300

TransferUsername="" TransferPassword="" TransferSrvHostname=“192.168.3.4"
Configuration of a User Password

If you are using the CLI, the new password will be used the next time you connect to the Aastra.
Aaa.Users[UserName=public].Password=TestPwd
Debugging
The following sections allow you to enable two useful debugging tools of the Aastra: syslog messages and
PCM traces.
Enabling Syslog
This example assumes that you run a syslog server at address 192.168.3.4.

Examples Software Configuration Guide
Nlm.SyslogRemoteHost=”192.168.3.4”
Cli.MinSeverity=300
Bni.MinSeverity=400
Hoc.MinSeverity=500
Configuring PCM Capture

The PCM traces are two different RTP streams made specifically to record all analog signals that are either
sent or received on the analog side of the Aastra. Only the configured port, port #1 and/or #2 are sending the
PCM traces for a maximum of four simultaneous RTP streams.
The RTP streams are sent to a configurable IP address, normally an IP address on your network where it can
be recorded with a packet sniffer (such as Wireshark). Moreover, they are independent from the regular RTP
streams of the VoIP call.
All streams are sent instantly at startup with an average ptime of 15 ms. This means that until the PCM traces
are disabled, even an idle unit will continuously send up to 66.6 packets/s X 4 streams = 267 packets/s using
approximately 174 bytes each, for a total of 46 Kbytes of upstream bandwidth.
Mipt.PcmCaptureEnable=1
Mipt.PcmCaptureEndpoint=”Bri1-1”
Mipt.PcmCaptureIpAddr=”192.168.3.3”
Mipt.restart

Appendix B - Scripting Language Examples

A P P E N D I X
C Maximum Transmission Unit

(MTU)
This appendix describes the MTU (Maximum Transmission Unit) requirements of the Aastra.
What is MTU?
The Maximum Transmission Unit (MTU) is a parameter that determines the largest packet than can be
transmitted by an IP interface (without it needing to be broken down into smaller units). Each interface used
by TCP/IP may have a different MTU value specified.
The MTU should be larger than or equal to the largest packet you wish to transmit unfragmented. Note that
this only prevents fragmentation locally. Some other link in the path may have a smaller MTU: the packet will
be fragmented at that point, although some routers may refuse packets larger than their MTU.
Aastra’s MTU
The Aastra’s MTU is 1500 bytes, which is the Ethernet typical value.
Possible Hardware Problem
The implementation of the IEEE Standard 802.1q in the Aastra may have a minor problem because of
hardware limitations.
802.1q increases the Ethernet frame header by 4 bytes, adding a Virtual LAN ID and a user_priority. This is
useful to limit broadcasts that cross bridges, and it may also prioritize frames in the queuing algorithm of
switches. However, it also increases the maximum possible size of Ethernet frames from 1518 to 1522 bytes,
and this might not be handled adequately by every hardware.
A workaround is available for PCs running Windows to avoid sending 1522 bytes packets (note that this
happens only in special and rare cases). The workaround is to reduce the MTU of the interface (the one that
sends packets with 802.1q framing) by 4 bytes.
1. Use the registry editor (regedt32) and go to the key:
Windows 2000 and later:
HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\Tcpip\Parameters\Interfaces
\<ethernet adapter>
Windows NT4 and 98:
\HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\<ethernet
adapter>\Parameters\Tcpip
where <Ethernet adapter> can be found by using the command “ipconfig /all”.
2. Add (or modify) a value named MTU of type REG_DWORD. Set it to 1496 (instead of 1500), in
decimal. Restart the computer to have those changes in effect.
In Windows 2000 and later this value is under the following key:
• Key: Tcpip\Parameters\Interfaces\ID for Adapter2

Appendix C - Maximum Transmission Unit (MTU) Possible Hardware Problem
• Value Type: REG_DWORD Number

• Valid Range: 68 - the MTU of the underlying network
• Default: 0xFFFFFFFF
• Description: This parameter overrides the default MTU for a network interface. The
MTU is the maximum packet size in bytes that the transport will transmit over the
underlying network. The size includes the transport header. Note that an IP datagram
may span multiple packets. Values larger than the default for the underlying network
will result in the transport using the network default MTU. Values smaller than 68 will
result in the transport using an MTU of 68.
3. To validate that the changes are correct, try to ping the Aastra with large packets once restarted:
ping -l 2000
This will cause IP fragmentation, the first fragment being as large as the interface allows it. With the
MTU reduced, you should now receive an answer. For more informations, see:
http://support.microsoft.com/default.aspx?scid=kb;en-us;120642.

A P P E N D I X
D Web Interface – SNMP Variables

Mapping
All parameters available in the Aastra web interface may also be configured via SNMP. The Aastra SNMP
feature offers the following options:
 Password-protected access
 Remote management
 Simultaneous management
This Appendix lists the mapping between the web interface fields and the corresponding SNMP variables of
the Aastra.
System Page
Information Sub-Page
Current Status Section
Field Name SNMP Variable Description
System Description unitInfoProductName Product name of the unit.
Firmware mfpInstalledInfoMfpVersion Version of the Firmware Pack installed.
Profile mfpInstalledInfoMfpProfileName Name of the profile.
MAC Address unitInfoMacAddress MAC address of the unit.
Serial Number unitInfoSerialNumber Serial number of the unit.
System Uptime sysUpTime Time since the last restart.
System Time currentTimeSystem Current date and time system configured in the unit.
Services Sub-Page
System Service
System Service servicesInfoName Current service name.
Status servicesInfoExecState Shows the execution state of the service.
User Service
User Service servicesInfoName Current service name.
Status servicesInfoExecState Shows the execution state of the service.
Startup Type servicesInfoStartupType Selects the service startup type.
Action serviceCommandsRestart Restarts, starts or stops the service.

Appendix D - Web Interface – SNMP Variables Mapping System Page
Comment servicesInfoComment Comments on the service's current state.
Start (command) Starts the service.
Stop (command) Stops the service.
Restart (command) Restarts the service.
Restart Required Services
Graceful Delay (min) graceDelay The delay (in minutes) allowed for telephony calls to be all
completed.
CancelRestartRequiredServices (command) Cancels the restart during the grace delay period.
RestartRequiredServices (command) Restarts all required services
Hardware Sub-Page
Unit Configuration Section
Unit Configuration portsConfiguration Configures how each port provides a link interface.
Clock Reference physicalLinkClockMode A port can either generate the clocking for the line or accept
the clock from the line.
BRI Cards Configuration Section
Slot physicalLinkInterfaceName Identifies the interface.
Clock Reference physicalLinkClockMode A port can either generate the clocking for the line or accept
the clock from the line.
PRI Cards Configuration Section
Slot physicalLinkInterfaceName Identifies the interface.
Clock Reference physicalLinkClockMode Indicates the preferred synchronisation source to use for the
internal clock of this digital card.
Line Type physicalLinkLineCoding Defines the transmission encoding of bits.
Endpoints Sub-Page
Unit States Section
Administrative unitAdminState Indicates the current maintenance state of a unit.
Operational unitOpState The operational state of the unit reflects the unit's internal
state.

System Page Software Configuration Guide
Usage unitUsageState The usage state of the unit indicates its running state.
Action unitUnlock Allows to use a unit.

unitLock Gracefully disallows to use a unit.
unitForceLock Forcefully disallows to use a unit.
Endpoint States Section
Endpoint endpointEpId String that identifies an endpoint in other tables.
Administrative endpointAdminState Administrative state of an endpoint.
Operational endpointOpState Operational state of an endpoint.
Usage endpointUsageState Ru7ning state of an endpoint.
Initial Administrative endpointInitialAdminStateConfig Initial administrative state of an endpoint.
Action endpointUnlock Allows using the endpoint.

endpointLock Gracefully disallows using the endpoint.
endpointForceLock Forcefully disallows using the endpoint.
Administration Section
Disable Unit When No unitDisabledWhenNoGatewayReadyEnable Indicates if the unit operational state is automatically set to
Gateways Are In State disable when all signaling gateways are not ready.
Ready
Shutdown Endpoint endpointAutomaticShutdownEnable Indicates if an endpoint is physically shutdown when in the

When Operational 'idle-unusable' usage state.
State is 'Disable' And
Its Usage State Is 'idle-
unusable'
Syslog Sub-Page
Remote Host syslogRemoteHost Host name and port number of the device that archives log
entries.
Authentication, minSeverity (aaaMIB) Minimal Severity of Notification

Authorization and
Accounting (AAA)
Basic Network minSeverity (bniMIB) Minimal Severity of Notification

Interface (BNI)
Call Routing (CROUT) minSeverity (cRoutMIB) Minimal Severity of Notification
Certificate Manager minSeverity (certMIB) Minimal Severity of Notification

(CERT)
Command Line minSeverity (cliMIB) Minimal Severity of Notification

Interface (CLI)
Configuration Manager minSeverity (confMIB) Minimal Severity of Notification

(CONF)
Device Control minSeverity (dcmMIB) Minimal Severity of Notification

Manager (DCM)
DHCP (DHCP) minSeverity (dhcpMIB) Minimal Severity of Notification
Endpoint minSeverity (epAdmMIB) Minimal Severity of Notification

Administration
(EpAdm)

Appendix D - Web Interface – SNMP Variables Mapping System Page
Endpoint Services minSeverity (epServMIB) Minimal Severity of Notification

(EpServ)
Ethernet manager minSeverity (ethMIB) Minimal Severity of Notification

(Eth)
Firmware Pack minSeverity (fpuMIB) Minimal Severity of Notification

Updater (FPU)
Host Configuration minSeverity (hocMIB) Minimal Severity of Notification

(HOC)
Integrated Services minSeverity (isdnMIB) Minimal Severity of Notification

Digital Network (ISDN)
Local Quality Of minSeverity (lQosMIB) Minimal Severity of Notification

Service (LQOS)
Media IP Transport minSeverity (miptMIB) Minimal Severity of Notification

(MIPT)
Notifications and minSeverity (nlmMIB) Minimal Severity of Notification

Logging Manager
(NLM)
Plain Old Telephony minSeverity (potsMIB) Minimal Severity of Notification

System Lines service
(POTS)
Process Control minSeverity (pcmMIB) Minimal Severity of Notification

Manager (PCM)
Service Controller minSeverity (scmMIB) Minimal Severity of Notification

Manager (SCM)
SIP ALG (SipAlg) minSeverity (sipAlgMIB) Minimal Severity of Notification
SIP Endpoint (SipEp) minSeverity (sipEpMIB) Minimal Severity of Notification
Diagnostic Traces diagnosticTracesEnable Enables traces allowing the Technical Assistance Centre to
further assist in resolving some issues.
Filter diagnosticTracesFilter Filter applied to diagnostic traces.
Events Sub-Page
Activation eventsActivation Current activation state for this system event.
Criteria eventsCriteria Expression an event must match in order to apply the

specified action.
Service N/A N/A
Notification N/A N/A
Action eventsAction Action to apply to the system event if the criteria matches.
Config Status eventsConfigStatus Configuration status of the row.
InsertEvent (command) Inserts a row to the EventsTable
eventsDelete Deletes this row.
Local Log Sub-Page
Local Log Status Section
Maximum Number of LocalLogMaxNbEntries Maximum number of entries that the local log can contain.
Entries When adding a new entry while the local log is full, the
oldest entry is erased to make room for the new one.

Network Page Software Configuration Guide
Number of Error LocalLogNbErrorEntries Current number of error entries in the local log.
Entries
Number of Critical LocalLogNbCriticalEntries Current number of critical entries in the local log.
Entries
Local Log Entries Section
Local Time LocalTime Local date and time at which the log entry was inserted.
Format is YYYY-MM-DD HH:MM:SS.
Severity Severity Severity of the log entry.
Service Name ServiceTextkey Textual identifier of the service that issued the log entry.
Service Key ServiceNumkey Numerical identifier of the service that issued the log entry.
Message Key NotificationId Numerical identifier of the notification message.
Message Content Message The readable content of the log message.
Network Page
Status Sub-Page
Interfaces Status Section
Interface networkInterfaceStatusInterfaceName Network interface name.
Link networkInterfaceStatusLinkName Name of the link interface associated with the network
interface.
IP Address networkInterfacesStatusIpAddr Current address and network mask of the network interface.
Default Router networkInterfacesDefaultRouter Current default gateway of the network interface.
Connection Uptime networkInterfacesConnectionUptime The time, in seconds, for which this IP interface has been
connected.
Status uplinkInterfaceStatus Operational status of the Uplink network interface.
VLAN Override netorknterfacesStatusVlanOverrideEnable Indicates if the VLAN ID of the current network interface has
been overridden by the values received from the LLDP
protocol.
LLDP Status Section
Type remoteMediaPolicyStateAppType The type of application.
Vlan ID remoteMediaPolicyStateVlanId VLAN ID.
User Priority (802.1Q) remoteMediaPolicyStatePriority 802.1Q User Priority.
DiffServ (DSCP) remoteMediaPolicyStateDscp DSCP (DiffServ).
Policy Flag remoteMediaPolicyStatePolicyFlag Indicates if an Endpoint Device wants to explicitly advertise

that the network policy for a specific application type is
required but is currently unknown.
Tagged Flag remoteMediaPolicyStateTaggedFlag The Tagged flag.

Appendix D - Web Interface – SNMP Variables Mapping Network Page
Host Status Section
Automatic subnetsAutomaticConfigurationInterface The network interface that provides the automatic

Configuration Interface configuration (E.g.: DNS servers, NTP server, etc.) to this
subnet.
Host Name Configuration
Host Name domainNamesInfoSubnetName Name of the subnet.
Domain Name domainNamesInfoDomainName Indicates the subnet's current domain name.
Default Gateway Configuration
IPv4 Default Gateway defaultRoutersInfoDefaultRouter Indicates the subnet's current default gateway.
IPv6 Default Gateway defaultRoutersInfoDefaultRouter Indicates the subnet's current default gateway.
DNS Configuration
Primary DNS dnsServersInfoIpAddress1 Indicates the subnets' first DNS server.
Secondary DNS dnsServersInfoIpAddress2 Indicates the subnets' secondary DNS server.
Third DNS dnsServersInfoIpAddress3 Indicates the subnets' third DNS server.
Fourth DNS dnsServersInfoIpAddress4 Indicates the subnets' fourth DNS server.
SNTP Configuration
Primary SNTP Host sntpServersInfoHostName1 Indicates the subnets' first NTP server.
Secondary SNTP Host sntpServersInfoHostName2 Indicates the subnets' second NTP server.
Third SNTP Host sntpServersInfoHostName3 Indicates the subnets' third NTP server.
Fourth SNTP Host sntpServersInfoHostName4 Indicates the subnets' fourth NTP server.
Advanced IP Routes Section
# advancedIpRoutesStatusIPriority Unique identifier of the row in the table.
Source Address advancedIpRoutesStatusSourceAddress Source address[/mask] criteria used to match the rule.
Source Link advancedIpRoutesStatusSourceLink Source link criteria used to match the rule.
Forward To Network advancedIpRoutesStatusForwardToNetwork Network on which the packet is forwarded.
State advancedIpRoutesStatusStatus Status of the rule.
IPv4 Routes Section
Link ipRoutesStatus Link (interface) ID.
Destination ipRoutesStatus Destination IP address or network address.
Gateway ipRoutesStatus Specifies the gateway IP address.
Protocol ipRoutesStatus Identifies the entity that installed the route.
Firewall Section
# networkRulesStatusPriority Unique identifier of the row in the table.
Source Address networkRulesStatusSourceAddress Source address[/mask] criteria an incoming packet must

have to match this rule.

Source Port networkRulesStatusSourcePort Source port[-port] criteria an incoming packet must have to
match this rule.
Destination Address networkRulesStatusDestinationAddress Destination address[/mask] criteria an incoming packet must

Destination Port networkRulesStatusDestinationPort Destination port[-port] criteria an incoming packet must have
to match this rule.
Protocol networkRulesStatusProtocol Protocol criteria an incoming packet must have to match this
rule.
Connection State networkRulesStatusConnectionState Connection state associated with the incoming packet.
Action networkRulesStatusAction Action taken when this rule matches a packet.
Network Address Translation Section
# sNatRulesStatusPriority Unique identifier of the row in the table.

dNatRulesStatusPriority
Source Address sNatRulesStatusSourceAddress Source address[/mask] criteria an incoming packet must

dNatRulesStatusSourceAddress have to match this rule.
Source Port sNatRulesStatusSourcePort Source port[-port] criteria an incoming packet must have to
dNatRulesStatusSourcePort match this rule.
Destination Address sNatRulesStatusDestinationAddress Destination address[/mask] criteria an incoming packet must

dNatRulesStatusDestinationAddress have to match this rule.
Destination Port sNatRulesStatusDestinationPort Destination port[-port] criteria an incoming packet must have
dNatRulesStatusDestinationPort to match this rule.
Protocol sNatRulesStatusProtocol Protocol criteria an incoming packet must have to match this
dNatRulesStatusProtocol rule.
New Address sNatRulesStatusNewAddress New address[:port] applied to the source of the packet.
dNatRulesStatusNewAddress
Host Sub-Page
General Configuration Section
Automatic automaticConfigurationInterface The network interface that provides the automatic

Configuration Interface configuration used by the unit (e.g.: Default gateway, DNS
servers, NTP server, etc.).
Automatic IPv4 config automaticConfigurationInterface The network interface that provides the automatic
source network: configuration used by the unit (e.g.: Default gateway, DNS
servers, NTP server, etc.).
Automatic IPv6 config ipv6AutomaticConfigurationInterface The network interface that provides the IPv6 automatic
source network configuration (Default Router, domain name, DNS servers
and NTP server) used by the unit.
Host Name Configuration Section
Domain Name domainNameConfigSource Configuration source for the domain name.

Configuration Source
Domain Name staticDomainName Static domain name.
Host Name hostName System’s host name.

Default Gateway Configuration Section
IPv4
Configuration Source defaultRouterConfigSource Configuration source for the default gateway.
Default Gateway staticDefaultRouter Static default gateway address.
IPv6
Configuration Source defaultRouterConfigSource Configuration source for the default gateway.
Default Gateway staticDefaultRouter Static default gateway address.
DNS Configuration Section
Configuration Source dnsServersConfigSource Configuration source for the DNS servers.
Primary DNS staticDnsServersIpAddress1 Indicates the subnets' first DNS server.
Secondary DNS staticDnsServersIpAddress2 Indicates the subnets' secondary DNS server.
Third DNS staticDnsServersIpAddress3 Indicates the subnets' third DNS server.
Fourth DNS staticDnsServersIpAddress4 Indicates the subnets' fourth DNS server.
SNTP Configuration Section
Configuration Source sntpConfigSource Configuration source for the SNTP parameters.
Primary SNTP staticSntpServersHostName1 Indicates the subnets' first NTP server.
Secondary SNTP staticSntpServersHostName2 Indicates the subnets' second NTP server.
Third SNTP staticSntpServersHostName3 Indicates the subnets' third NTP server.
Fourth SNTP staticSntpServersHostName4 Indicates the subnets' fourth NTP server.
Synchronization Period sntpSynchronizationPeriod Time interval between system time synchronization cycles.
Synchronization Period sntpSynchronizationPeriodOnError Time interval between retries after an unsuccessful request
On Error to the SNTP server.
Time Configuration Section
Static Time Zone staticTimeZone Specifies the time zone in which the system is located.
Interfaces Sub-Page
Interface Configuration Section
Interface networkInterfacesInterfaceName Network interface name.
Link networkInterfacesLinkName Name of the link interface associated with the network
interface.
Type networkInterfacesConnectionType Connection type of the network interface.
Static IP Address networkInterfacesStaticIpAddr IPv4 address and network mask of the network interface.

Static Default Router networkInterfacesStaticDefaultRouter IPv4 address of the default gateway for the network
interface when the ConnectionType is set to ipStatic.
Activation networkInterfacesActivation Attempts to activate the network interface.
AddNetwork (command) Adds a new network interface.
networkInterfacesDelete Deletes the network interface and removes it from the

system.
PPPoE Configuration Section
Service Name pppServiceName Name of the service requested to the access concentrator
when establishing the next PPPoE connection.
Protocol pppAuthenticationProtocol Authentication protocol to use for authenticating the system

to the PPP peer.
User Name pppIdentity Name that identifies the system to the PPP peer during the
authentication process.
Password pppSecret Secret that identifies the system to the PPP peer during the
authentication process.
LLDP Configuration Section
Network Interface NetworkInterface The network interface name on which LLDP should be
enabled.
Chassis ID ChassisId The address type to populate the chassis ID.
Override Network OverrideNetworkPolicyEnable Enables the LLDP-MED protocol override of the VLAN ID,
Policy User Priority and DiffServ values.
Ethernet Link Configuration Section
Link linksName The name of the Ethernet link.
MTU linksMtu Configures the MTU (Maximum Transmission Unit) of a

specific Ethernet link.
802.1x Authentication linksIeee8021XAuthentication Configures the IEEE 802.1x authentication protocol

activation on the Ethernet link interface.
EAP Username eapUserName Username used to authenticate each Ethernet link

interfaces during the IEEE 802.1x EAP-TLS authentication
process.
Certificate Validation eapCertificationValidation Level of validation used by the device to authenticate the
IEEE 802.1x EAP-TLS peer's certificate. This variable
controls also the criteria used to select the host certificate
sent during the authentitication handshake
VLAN Sub-Page
Link vlanLinkName Name of the Ethernet link over which the VLAN interface is
built.
Id vlanId VLAN ID used by the VLAN interface.
Default User Priority vlanDefaultUserPriority Default User Priority value the interface uses when tagging
packets.

AddVlan (command) Adds a new virtual LAN.
vlanDelete Deletes the VLAN interface and removes it from the system.
Local Firewall Sub-Page

Config Modified configModifiedStatus Shows whether the configuration of the local firewall was
modified without being applied.
Local Firewall Configuration Section
Default Policy defaultPolicy Action taken when a packet doesn't match any rules.
Local Firewall Rules Section
# localRulesPriority Unique identifier of the row in the table.
Activation localRulesActivation Current state for this rule.
Source Address localRulesSourceAddress Source address of the incoming packet.
Source Port localRulesSourcePort Source port of the incoming packet.
Destination Address localRulesDestinationAddress Destination address of the incoming packet.
Destination Port localRulesDestinationPort Destination port of the incoming packet.
Protocol localRulesProtocol Protocol of the incoming packet.
Action localRulesAction Action that will be taken in the matching packets.
localRulesInsert Inserts a new row before this row.
localRulesDelete Deletes this row.
localRulesDown Moves the current row downside.
localRulesUp Moves the current row upside.
IP Routing Sub-Page
Config Modified configModifiedStatus Shows whether or not the Network Address Translation
configuration has been modified without being applied.
Rollback (command) Rolls back the current configuration to the running

configuration as showed in the status.
IP Routing Configuration Section
IPv4 Forwarding ipv4ForwardingEnable Enables/disables IPv4 forwarding.

Advanced IP Routes Section
# advancedIpRoutesPriority Unique identifier of the row in the table.
Activation advancedIpRoutesActivation Activates this route.
Source Address advancedIpRoutesSourceAddress Specifies the source IP address criteria an incoming packet
must have to match this rule.
Source Link advancedIpRoutesSourceLink Specifies the source link criteria an incoming packet must
Forward to Network advancedIpRoutesForwardToNetwork Network on which to route the packet.
advancedIpRoutesInsert Inserts a new row before this row.
advancedIpRoutesDelete Deletes this row.
advancedIpRoutesDown Moves the current row downside.
advancedIpRoutesUp Moves the current row upside.
Static IP Routes Section
Index staticIpRoutesIndex Unique identifier of the row in the table.
Destination staticIpRoutesDestination Specifies the destination IP address criteria that an outgoing

packet must have to match this route.
Link staticIpRoutesLink Output link (interface) name.
Gateway staticIpRoutesGateway Specifies the IP address of the gateway used by the route.
insertStaticIpRoute (command) Inserts a new row at the end of the StaticIpRoutes table.
staticIpRoutesDelete Deletes this row.
Network Firewall Sub-Page

Config Modified configModifiedStatus Shows whether the configuration of the network firewall was

Network Firewall Configuration Section
Default Policy defaultPolicy Action taken when a packet does not match any rules.
Network Firewall Rules Section
# networkRulesPriority Unique identifier of the row in the table.
Activation networkRulesActivation Activates this rule.
Source Address networkRulesSourceAddress Source address of the incoming packet.
Source Port networkRulesSourcePort Source port of the incoming packet.

Destination Address networkRulesDestinationAddress Destination address of the incoming packet.
Destination Port networkRulesDestinationPort Destination port of the incoming packet.
Protocol networkRulesProtocol Protocol of the incoming packet.
Connection State networkRulesConnectionState Connection state associated with the incoming packet.
Action networkRulesAction Action that will be taken in the matching packets.
networkRulesInsert Inserts a new row before this row.
networkRulesDelete Deletes this row.
networkRulesDown Moves the current row downside.
networkRulesUp Moves the current row upside.
NAT Sub-Page
Config Modified configModifiedStatus Shows whether or not the Network Address Translation
configuration has been modified without being applied.

Source Network Address Translation Rules Section
# sNatRulesPriority Unique identifier of the row in the table.
Activation sNatRulesActivation Activates this rule.
Source Address sNatRulesSourceAddress Source address of the incoming packet.
Source Port sNatRulesSourcePort Source port of the incoming packet.
Destination Address sNatRulesDestinationAddress Destination address of the incoming packet.
Destination Port sNatRulesDestinationPort Destination port of the incoming packet.
Protocol sNatRulesProtocol Protocol of the incoming packet.
New Address sNatRulesNewAddress New address applied to the destination of the packet.
sNatRulesInsert Inserts a new row before this row.
sNatRulesDelete Deletes this row.
sNatRulesDown Moves the current row downside.
sNatRulesUp Moves the current row upside.
Destination Network Address Translation Rules Section
# dNatRulesPriority Unique identifier of the row in the table.
Activation dNatRulesActivation Activates this rule.
Source Address dNatRulesSourceAddress Source address of the incoming packet.
Source Port dNatRulesSourcePort Source port of the incoming packet.
Destination Address dNatRulesDestinationAddress Destination address of the incoming packet.

Destination Port dNatRulesDestinationPort Destination port of the incoming packet.
Protocol dNatRulesProtocol Protocol of the incoming packet.
New Address dNatRulesNewAddress New address applyed to the destination of the packet.
dNatRulesInsert Inserts a new row before this row.
dNatRulesDelete Deletes this row.
dNatRulesDown Moves the current row downside.
dNatRulesUp Moves the current row upside.
DHCP Server Sub-Page
DHCP Server Status Section
Subnet Status subnetsConfigStatus Subnet configuration status.
Lease Time leaseTimesInfoDefault Indicates the subnet's current default lease time in seconds.
Default Gateway defaultRoutersInfoDefaultRouter Indicates the subnet's current default gateway.
DNS (Option 6)
Primary DNS dnsServersInfoDns1 Indicates the subnets' first DNS server.
Secondary DNS: dnsServersInfoDns2 Indicates the subnets' secondary DNS server.
Third DNS dnsServersInfoDns3 Indicates the subnets' third DNS server.
Fourth DNS dnsServersInfoDns4 Indicates the subnets' fourth DNS server.
NTP (Option 42)
Primary SNTP Host ntpServersInfoNtp1 Indicates the subnets' first NTP server.
Secondary SNTP Host ntpServersInfoNtp2 Indicates the subnets' second NTP server.
Third SNTP Host ntpServersInfoNtp3 Indicates the subnets' third NTP server.
Fourth SNTP Host ntpServersInfoNtp4 Indicates the subnets' fourth NTP server.
DHCP Server Leases Section
MAC Address assignedLeasesInfoMacAddress MAC address of the host.
IP Address assignedLeasesInfoIpAddress IP Address of the host.
Subnet Name assignedLeasesInfoSubnetName Indicates on which subnet the host is located.
Time Left assignedLeasesInfoLeaseTimeLeft Indicates the lease time left in seconds.
DHCP Server Configuration Section
DHCP Server Enable subnetsEnableSubnet Enables the subnet configuration.
Start IP Address subnetsStartAddress Start address of the subnet range.

End IP Address subnetsEndAddress End address of the subnet range.
Automatic subnetsAutomaticConfigurationInterface Interface that will provide the automatic configuration to this
Configuration Interface subnet.
Subnet Specific specificLeaseTimesEnableConfig Defines the lease time configuration to use for a specific
subnet.
Lease Time defaultLeaseTime Specifies the lease time (in seconds) default setting for all
specificLeaseTimesLeaseTime subnets.
Specifies the subnet's specific lease time in seconds.
Domain Name (Option 15)
Enable Option specificDomainNamesEnableOption Enables the domain name option.
Subnet Specific specificDomainNamesEnableConfig Defines the domain name configuration to use for this
specific subnet.
Configuration Source defaultDomainNameConfigSource Default configuration source of all subnets.

specificDomainNamesConfigSource Subnet's domain name specific configuration source.
Domain Name defaultStaticDomainName Default static domain name for all subnets.
specificDomainNamesStaticName Static Domain Name Configuration.
Enable Option specificDefaultRoutersEnableOption Enables the default gateway option.
Configuration Source specificDefaultRoutersConfigSource The subnet's specific router configuration source.
Default Gateway specificDefaultRoutersStaticRouter Specifies the subnet's default gateway.
DNS (Option 6)
Enable Option specificDnsServersEnableOption Enables the DNS servers option.
Subnet Specific specificDnsServersEnableConfig Defines the DNS servers configuration to use for a specific
subnet.
Configuration Source defaultDnsServersConfigSource Default configuration source for the DNS servers of all
specificDnsServersConfigSource subnets.
DNS servers specific configuration source for the subnet.
Primary DNS specificDnsServersStaticDns1 IP address of the first DNS server of the subnet.
Secondary DNS: specificDnsServersStaticDns2 IP address of the second DNS server of the subnet.
Third DNS specificDnsServersStaticDns3 IP address of the third DNS server of the subnet
Fourth DNS specificDnsServersStaticDns4 IP address of the fourth DNS server of the subnet.
NTP (Option 42)
Enable Option specificNtpServersEnableOption Enables the NTP servers option.
Subnet Specific specificNtpServersEnableConfig Defines the NTP servers configuration to use for a specific
subnet.
Configuration Source defaultNtpServersConfigSource Default configuration source for the NTP servers of all
specificNtpServersConfigSource subnets.
NTP servers specific configuration source for the subnet.
Primary NTP specificNtpServersStaticNtp1 IP address of the first NTP server of the subnet.
Secondary NTP specificNtpServersStaticNtp2 IP address of the second NTP server of the subnet.
Third NTP specificNtpServersStaticNtp3 IP address of the third NTP server of the subnet.
Fourth NTP specificNtpServersStaticNtp4 IP address of the fourth NTP server of the subnet.
NBNS (Option 44)
Enable Option specificNbnsServersEnableOption Enable NBNS servers option.
Subnet Specific specificNbnsServersEnableConfig Defines the NBNS servers configuration to use for a specific
subnet.
Primary NBNS specificNbnsServersStaticNbns1 IP address of the first NBNS server of the subnet.
Secondary NBNS specificNbnsServersStaticNbns2 IP address of the second NBNS server of the subnet.

POTS Page Software Configuration Guide
Third NBNS specificNbnsServersStaticNbns3 IP address of the third NBNS server of the subnet.
Fourth NBNS specificNbnsServersStaticNbns4 IP address of the fourth NBNS server of the subnet.
QoS Sub-Page
Differentiated Service Field Configuration Section
Default DiffServ (IPv4) defaultDiffServ Default Differentiated Services value used by the unit for all
generated packets.
Default Traffic Class defaultTrafficClass Default Traffic Class value used by the unit for all generated
(IPv6) IPv6 packets.
Ethernet 802.1Q Tagging Configuration Section

Enable ethernet8021QTaggingEnablePriorityTagging Enables or disables user priority tagging on the interface.
Default User Priority ethernet8021QTaggingDefaultUserPriority Default User Priority value the interface uses when tagging
packets.
Service Class Configuration Section

DiffServ (IPv4) serviceClassesDiffServ Differentiated Services value for a specific service class.
Traffic Class (IPv6) serviceClassesTrafficClass Default Traffic Class value used in IPv6 packets.
User Priority servicesClassesUserPriority User priority for a specific service class.
Network Traffic Control Configuration Section

Physical Link linkBandwidthControlLinkName Name of the Ethernet link over which the bandwidth
limitation is applied.
Egress Limit linkBandwidthControlEgressLimit Indicates the bandwidth limitation for the selected link
interface.
POTS Page
Status Sub-Page
Line Status Section
ID lineId String that identifies a line in other tables.
Type lineTypeStatus The status POTS type of the line.
State lineState The current call control state for this channel.
Config Sub-Page

Appendix D - Web Interface – SNMP Variables Mapping POTS Page
Caller ID CallerIdCustomization Allows selecting the detection/generation method of caller

Customization ID.
Caller ID Transmission CallerIdTransmission Allows selecting the transmission type of the caller ID.
Vocal Unit Information VocalUnitInformation Determines whether or not the unit's IP or MAC address or
firmware version number can be acquired using the *#*0,
*#*1, and *#*8 digit maps respectively.
FXS Config Sub-Page
FXS Configuration Section
Line Supervision Mode fxsLineSupervisionMode Determines how the power drop and line polarity are used to
signal the state of a line.
Disconnect Delay fxsDisconnectDelay Determines whether or not call clearing occurs as soon as
the called user is the first to hang up a received call.
Auto Cancel Timeout fxsDefaultAutoCancelTimeout Time, in seconds, the endpoint rings before the call is
automatically cancelled.
Inband Ringback fxsInbandRingback Determines whether or not the FXS endpoint needs to
generate a ringback for incoming ringing call.
Shutdown Behavior fxsShutdownBehavior Determines the FXS endpoint behavior when it becomes
shut down.
Power Drop on fxsPowerDropOnDisconnectDuration Determines the power drop duration that is made at the end
Disconnect Duration of a call when the call is disconnected by the remote party.
Service Activation FxsServiceActivation Selects the method used by the user to activate
supplementary services like call hold, second call, call
waiting, call transfer and conference call.
FXS Country Configuration Section
Override Country fxsCountryCustomizationOverride Allows overriding FXS-related default country settings.

Customization
Country Override Loop fxsCountryCustomizationLoopCurrent Loop current generated by the FXS port in ma.
Current
Country Override Flash fxsCountryCustomizationFlashHookDetectionR The range in which the hook switch must remain pressed to
Hook Detection Range ange perform a flash hook.
FXS Bypass Section
Endpoint fxsBypassId String that identifies a line in other tables.
Activation fxsBypassActivation Specifies when the bypass needs to be activated.
Activation DTMF Map fxsBypassActivationDtmfMap Specifies the DTMFs to signal to enable the bypass.
Deactivation Timeout fxsBypassDeactivationTimeout Specifies the delay to wait before deactivating the bypass
after an on hook if the bypass is activated on demand.

SIP Page Software Configuration Guide
SIP Page
Gateways Sub-Page
SIP Gateway Status Section
Name gatewayStatusName Name of the SIP gateway.
Network Interface gatewayStatusNetworkInterface Network on which the gateway listens for incoming SIP
traffic.
Port gatewayStatusPort Port on which the gateway listens for incoming unsecure
SIP traffic.
Secure Port gatewayStatusSecurePort Port on which the gateway listens for incoming secure SIP
traffic.
State gatewayStatusState Current state of the gateway.
SIP Gateway Configuration Section
Name gatewayName Name of the SIP gateway. It identifies the gateway in other
tables.
Network Interface gatewayNetworkInterface Network on which the gateway listens for incoming SIP
traffic.
Port gatewayPort Port on which the gateway listens for incoming unsecure
SIP traffic.
Secure Port gatewayStatusSecurePort Port on which the gateway listens for incoming secure SIP
traffic.
InsertGateway (command) Adds a row.
gatewayDelete Deletes this row.
Servers Sub-Page
RegistrationRefresh (command) Command to refresh the registrations.
TLS Persistent Connections Status Section
Gateway tlsPersistentConnectionStatusGateway The SIP gateway used to register.
Local Port tlsPersistentConnectionStatusLocalPort Local port used by the TLS persistent connection.
Configured Remote tlsPersistentConnectionStatusRemoteHost The remote host used to establish the TLS persistent
Host connection.
Remote IP Address tlsPersistentConnectionStatusRemoteAddress The resolved IP address of the remote host used to
establish the TLS persistent connection.
State tlsPersistentConnectionStatusState The current state of the TLS persistent connection.

Appendix D - Web Interface – SNMP Variables Mapping SIP Page
SIP Default Servers Section
Registrar Host defaultStaticRegistrarServerHost SIP registrar server FQDN and port.
Proxy Host defaultStaticProxyHomeDomainHost SIP proxy server FQDN and port.
Outbound Proxy Host defaultStaticProxyOutboundHost SIP outbound proxy server FQDN and port.
Messaging Server defaultStaticMessagingHost Messaging server FQDN and port.

Host
SIP Gateway Specific Registrar Servers Section
Gateway Name gwSpecificRegistrationGatewayName String that identifies a SIP gateway in other tables.
Gateway Specific gwSpecificRegistrationEnableConfig Defines the configuration to use for a specific SIP gateway.
Registrar Host gwSpecificRegistrationServerHost SIP registrar server FQDN and port for a specific SIP
gateway.
SIP Gateway Specific Proxy Servers Section
Gateway Name gwSpecificProxyGatewayName String that identifies a SIP gateway in other tables.
Gateway Specific gwSpecificProxyEnableConfig Defines the configuration to use for a specific SIP gateway.
Proxy Host gwSpecificProxyHomeDomainHost SIP proxy server FQDN and port for a specific SIP gateway.
Outbound Proxy Host gwSpecificProxyOutboundHost SIP outbound proxy server FQDN and port for a specific SIP
gateway.
Keep Alive Section
Keep Alive Method sipKeepAliveMethod Method used to perform the SIP keep alive.
Keep Alive Interval sipKeepAliveInterval Defines the interval, in seconds, at which SIP OPTIONS are
sent to verify the server status.
Keep Alive Destination sipKeepAliveDestination Determines the behaviour of the device when performing
the keep alive action.
SIP Gateway Specific Keep Alive Targets Section
Gateway Name gwKeepAliveAlternateDestinationGatewayNam String that identifies a SIP gateway in other tables.
e
Alternate Target gwKeepAliveAlternateDestinationAlternateDesti Alternate destination target server FQDN and port for a
nation specific SIP gateway.
Registrations Sub-Page
Endpoints Registration Status Section
Endpoint registrationStatusEndpoint The endpoint related to this registration.
User Name registrationStatusUsername The username currently used by the registration.

Gateway Name registrationStatusGateway The SIP gateway used to register.
Registrar registrationStatusRegistrar The host of the registrar currently used by the registration.
Status registrationStatusState The current state of the registration.
Endpoints Messaging Subscription Status Section
Endpoint mwiStatusEndpoint The endpoint related to this subscription.
User Name mwiStatusUsername The username currently used by the subscription.
Gateway Name mwiStatusGatewayName The SIP gateway used for this subscription.
Messaging Host mwiStatusMessagingHost Messaging server FQDN and port used to subscribe the
event state.
MWI Status mwiStatusSubscriptionState The current state of the subscription.
Unit Registration Status Section
User Name registrationStatusUsername The username currently used by the registration.
Gateway Name registrationStatusGateway The SIP gateway used to register.
Registrar registrationStatusRegistrar The host of the registrar currently used by the registration.
Status registrationStatusState The current state of the registration.
Endpoints Registration Section
Endpoint userAgentEpId String that identifies an endpoint in other tables.
User Name userAgentUserName String that uniquely identifies this endpoint in the domain.
Friendly Name userAgentFriendlyName Friendly name for SIP User Agent.
Register userAgentRegister Indicate whether the endpoint needs to register to the

registrar.
Gateway Name userAgentGatewayName Selects on which SIP gateway the user configuration is
applied.
Unit Registration Section
Index registrationUsersIndex Unique identifier of the row.
User Name registrationUsersUsername String that uniquely identifies this user in the domain.
Gateway Name registrationUsersGatewayName Selects on which SIP gateway the user configuration is
applied.
registrationInsertUser (command) Adds a row.
registrationUsersDelete Delete this row.

Registration Configuration Section
Default Registration defaultRegistrationRefreshTime Defines the time, relative to the end of the registration, at
Refresh Time which a registered unit will begin updating its registration.
Proposed Expiration defaultRegistrationProposedExpirationValue Configures the suggested expiration delay of a contact in

Value In Registration the SIP REGISTER.
Default Expiration defaultRegistrationExpirationValue Configures the default registration expiration.

Value In Registration
Authentication Sub-Page
Index authenticationIndex Authentication index for this row.
Apply to authenticationApplyTo Entity to which apply authentication.
Endpoint authenticationEpId Endpoint Identification.
Gateway authenticationGatewayName String that identifies a SIP gateway in other tables.
Validate Realm authenticationValidateRealm Defines whether or not the current credentials are valid for
any realm.
Realm authenticationRealm Authentication Realm.
User Name authenticationUserName String that uniquely identifies this entity in the realm.
Password authenticationPassword User password.
authenticationInsert Inserts a new row before this row.
authenticationDelete Deletes this row.
authenticationDown Moves the current row downside.
authenticationUp Moves the current row upside.
Transport Sub-Page
Add SIP Transport in transportConfigRegistrationEnable Indicates whether or not the SIP Gateway must include its
Registration supported transports in its registrations.
Add SIP Transport in transportConfigContactEnable Indicates whether or not the SIP Gateway must include its
Contact Header supported transport in all SIP messages that have the
contact header, except for the REGISTER message.
Persistent TLS Base transportTlsPersistentBasePort Base port used to establish TLS persistent connections with
Port SIP servers when the TLS transport is enabled.
Persistent TLS Retry transportTlsPersistentRetryInterval Time interval before retrying the establishment of a TLS
Interval persistent connection.
TLS Trusted Certificate transportTlsCertificateTrustLevel Defines how a peer certificate is considered trusted for a
Level TLS connection.
TCP Connect Timeout interopTcpConnectTimeout Defines the maximum time, in seconds, the unit should try to
establish a TCP or TLS connection to SIP hosts.

Protocol Configuration Section
UDP transportConfigUdpEnable Enables or disables the UDP transport.
UDP QValue transportConfigUdpQValue Indicates the priority of the UDP transport.
TCP transportConfigTcpEnable Enables or disables the TCP transport.
TCP QValue transportConfigTcpQValue Indicates the priority of the TCP transport.
TLS transportConfigTlsEnable Enables or disables the TLS transport.
TLS QValue transportConfigTlsQValue Indicates the priority of the TLS transport.
Interop Sub-Page
Behavior on T.38 INVITE Not Accepted Section
SIP Error Code behaviorOnT38InviteNotAcceptedSipErrorCode SIP code in the error response to an INVITE for T.38 fax.
Behavior behaviorOnT38InviteNotAcceptedBehavior Behavior of the device when receiving a SIP error response
to an INVITE for T.38 fax.
SIP Interop Section
Secure Header interopSiemensTransportHeaderEnable Add the 'x-Siemens-Call-Type' header to the SIP packets.
Default Username interopDefaultUsernameValue Username to use when the username is empty or undefined.
Value
OPTIONS Method interopSipOptionsMethodSupport Determines the behaviour of the device when answering a
Support SIP OPTIONS request.
Ignore OPTONS on no InteropIgnoreSipOptionsOnNoUsableEndpoints Determines whether or not the SIP OPTIONS requests
usable endpoints should be ignored when all endpoints are unusable.
Behavior On Machine InteropBehaviorOnMachineDetection Specifies the SIP device behavior when a machine is
Detection detected during a call.
Registration Contact InteropRegistrationContactMatching Specifies the matching behaviour for the contact header
Matching received in positive responses to REGISTER requests sent
by the unit.
Transmission Timeout interopTransmissionTimeout Changes the time to wait for a response or an ACK before
considering a transaction timed out.
SDP Interop Section
Offer Answer Model
Answer Codec answerCodecNegotiation Defines the codec negotiation rule when generating a SDP
Negotiation answer.
Enforce Offer Answer interopEnforceOfferAnswerModel Determines whether or not the unit requires strict adherence
Model to RFC 3264 from the peer when negotiating capabilities for
the establishment of a media session.
Allow Less Media in interopAllowLessMediaInResponse Selects whether or not the unit enables the mapping
Response between the "+" prefix of the username and the "type of
number" property.
Allow Media interopAllowMediaReactivationInAnswer Determines the unit behaviour when receiving a SDP
Reactivation in Answer answer activating a media that had been previously
deactivated in the offer.
Multiple Active Media

Allow Audio and Image interopAllowAudioAndImageNegotiation Determines the unit behaviour when offering media or
Negotiation answering to a media offer with audio and image
negotiation.
Allow Multiple Active interopAllowMultipleActiveMediaInAnswer Determines the behaviour of the device when answering a
Media in Answer request offering more than one active media.
Other
On Hold SDP Stream interopOnHoldSdpStreamDirection Define how to set the direction attribute and the connection
Direction in Answer address in the SDP when answering a hold offer with the
direction attribute “sendonly”.
Codec Vs Bearer interopCodecVsBearerCapabilitiesMappingPref Configures the behavior of the

Capabilities Mapping erredCodecChoice CodecVsBearerCapabilitiesMapping table by modifying the
Preferred Codec selection of the preferred codec in the incoming SDP.
Choice
TLS Interop Section
Certificate Validation InteropTlsCertificateValidation Specifies which level of security is used to validate the peer
certificate.
Misc Interop Section
Map Plus to TON interopMapPlusToTonInternational Defines the behaviour of the unit when receiving less media
International announcements in the response than in the offer.
Ignore Plus in interopIgnorePlusInUsername Determines whether or not the plus character (+) is ignored
Username when attempting to match a challenge username with
usernames in the Authentication table.
Escape Pound (#) in interopEscapePoundInSipUriUsername Determines whether or not the pound character (#) must be
SIP URI Username escaped in the username part of a SIP URI.
Escape Format interopEscapeFormat Configures the escaped characters to lower or uppercase

hexadecimal format in SIP headers.
Misc Sub-Page
Penalty Box Section
Penalty Box Activation penaltyBoxEnable Indicates whether the unit uses the penalty box feature.
Penalty Box Time penaltyBoxTime Amount of time that a host spends in the penalty box.
SIP to Cause Error Mapping Section
SIP Code errorMappingSipToCauseSipCode SIP code to map to a cause.
Cause errorMappingSipToCauseCause Cause to map to the SIP code.
ErrorMappingInsertSipToCause (command) Inserts a new row before this row.
erromappingSipToCauseDelete Deletes this row.

Configure New SIP To Cause Error Mapping Panel
SIP Code errorMappingSipToCauseSipCode SIP code to map to a cause.
Cause errorMappingSipToCauseCause Cause to map to the SIP code.
Cause to SIP Error Mapping
Cause errorMappingCauseToSipCause Cause to map to the SIP code.
SIP Code errorMappingCauseToSipSipCode SIP code to map to a cause.
ErrorMappingInsertCauseToSip (command) Inserts a new row before this row.
erromappingCauseToSipDelete Deletes this row.
Cause To SIP Error Mapping Panel
Cause errorMappingCauseToSipCause Cause to map to the SIP code.
SIP Code errorMappingCauseToSipSipCode SIP code to map to a cause.
Additional Headers Section
Reason Header ReasonHeaderSupport Indicates whether or not the unit uses the SIP reason
Support header.
Referred-By Support ReferredByHeader Indicates how the Referred-By header is used when
participating in a transfer.
PRACK Section
UAS PRACK Support uasPrackSupport Determines the support of RFC 3262 (PRACK) when acting
as as user agent server.
UAC PRACK Support uacPrackSupport Determines the support of RFC 3262 (PRACK) when acting
as as user agent client.
Session Refresh Section
Session Refresh Timer defaultSessionTimerEnable Enables/Disables the session expiration services.

Enable
Minimum Expiration defaultSessionTimerMinimumExpirationDelay Minimum value for the periodical session refreshes.
Delay (s)
Maximum Expiration defaultSessionTimerMaximumExpirationDelay Suggested maximum time for the periodical session
Delay (s): refreshes.
Session Refresh sessionRefreshRequestMethod Selects the method used for sending Session Refresh
Request Method Requests.

Appendix D - Web Interface – SNMP Variables Mapping Media Page
SIP Gateway Configuration Section
Gateway Name gatewayName Name of the SIP gateway. It identifies the gateway in other
tables.
SIP Domain Override gatewayDomain Controls whether or not to override the SIP domain used.
SIP Transfer Section
Blind Transfer Method BlindTransferMethod Selects the SIP method to use in a blind transfer scenario.
Diversion Section
Method diversionConfigMethod Selects the SIP method used to receive/send call diversion
information in an INVITE.
Event Handling Section
Gateway Name gwEventHandlingGatewayName String that identifies a SIP gateway in other tables.
Reboot gwEventHandlingReboot Specifies whether a remote reboot via a SIP NOTIFY

message event is supported or not for a specific SIP
gateway.
CheckSync gwEventHandlingCheckSync Specifies whether a transfer script via a SIP NOTIFY

message event is supported or not for a specific SIP
gateway.
Messaging Subscription Section
Username in Request- defaultUsernameInRequestUriEnable Indicates whether or not the unit adds the username in the
URI gwSpecificMwiUsernameInRequestUriEnable request URI of MWI SUBSCRIBE requests.
Advice Of Charge (AOC) Section
Gateway Name GatewayName String that identifies a SIP gateway in other tables.
AOC-D Support AocDSupport Specifies whether AOC (D)uring a call is supported for a
AOC-E Support AocESupport Specifies whether AOC at the (E)nd of a call is supported for
a specific SIP gateway.
Media Page

Media Page Software Configuration Guide
CODECS Sub-Page
Select Endpoint endpointEpId String that identifies an endpoint in other tables.
CODEC Section
epSpecificCodecG711AlawEnableConfig
epSpecificCodecG711MulawEnableConfig
epSpecificCodecG723EnableConfig
epSpecificCodecG726r16kbpsEnableConfig
Endpoint Specific Configuration to use for a specific endpoint.
epSpecificCodecG729EnableConfig
epSpecificCodecT38EnableConfig
epSpecificCodecClearModeEnableConfig
epSpecificCodecClearChannelEnableConfig
epSpecificCodecXCCDEnableConfig
defaultCodecG711AlawVoiceEnable
epSpecificCodecG711AlawVoiceEnable
defaultCodecG711MulawVoiceEnable
epSpecificCodecG711MulawVoiceEnable
defaultCodecG723VoiceEnable
epSpecificCodecG723VoiceEnable
defaultCodecG726r16kbpsVoiceEnable
epSpecificCodecG726r16kbpsVoiceEnable
defaultCodecG726r32kbpsVoiceEnable Indicates whether the codec can be selected for voice

Voice
epSpecificCodecG726r32kbpsVoiceEnable transmission.
defaultCodecG729VoiceEnable
epSpecificCodecG729VoiceEnable
defaultCodecClearModeVoiceEnable
epSpecificCodecClearModeVoiceEnable
defaultCodecClearChannelVoiceEnable
epSpecificCodecClearChannelVoiceEnable
defaultCodecXCCDVoiceEnable
epSpecificCodecXCCDVoiceEnable

defaultCodecG711AlawDataEnable
epSpecificCodecG711AlawDataEnable
defaultCodecG711MulawDataEnable
epSpecificCodecG711MulawDataEnable
defaultCodecG726r32kbpsDataEnable
epSpecificCodecG726r32kbpsDataEnable
defaultCodecG726r40kbpsDataEnable
epSpecificCodecG726r40kbpsDataEnable Indicates whether the codec can be selected for data
Data
transmission.
defaultCodecT38DataEnable
epSpecificCodecT38DataEnable
defaultCodecClearModeDataEnable
epSpecificCodecClearModeDataEnable
defaultCodecClearChannelDataEnable
epSpecificCodecClearChannelDataEnable
defaultCodecXCCDDataEnable
epSpecificCodecXCCDDataEnable
CODEC vs. Bearer Capabilities Mapping Section
Index defaultCodecVsBearerCapabilitiesMappingInde Index of the current Codec vs. Bearer match.

x
Enable defaultCodecVsBearerCapabilitiesMappingEna Defines if the outgoing codecs priority or selection should

bleMap reflect the incoming ITC and vice versa.
CODEC defaultCodecVsBearerCapabilitiesMappingCod The codec to be prioritized or selected in an outgoing

ec INVITE when the incoming SETUP's ITC matches
defaultCodecVsBearerCapabilitiesMappingInformationTran
sferCap.
Mapping Type defaultCodecVsBearerCapabilitiesMappingMap The ITC value to be set in the outgoing SETUP when the
pingType incoming INVITE's priority codec matches
defaultCodecVsBearerCapabilitiesMappingCodec.
ITC defaultCodecVsBearerCapabilitiesMappingInfor Mapping Type

mationTransferCap
Generic Voice Activity Detection (VAD)
Endpoint Specific epSpecificCodecEnableConfig Configuration to use for a specific endpoint.
Enable (G.711 and defaultCodecGenericVoiceActivityDetection Generic VAD configuration.

G.726) epSpecificCodecGenericVoiceActivityDetection
G.711 a-law Section
Use Endpoint Specific epSpecificCodecG711AlawEnableConfig Configuration to use for a specific endpoint.
Voice Transmission defaultCodecG711AlawVoiceEnable Indicates whether the codec can be selected for voice
epSpecificCodecG711AlawVoiceEnable transmission.
Voice Priority defaultCodecG711AlawVoicePriority Priority of this voice codec versus the other voice codecs.
epSpecificCodecG711AlawVoicePriority
Data Transmission defaultCodecG711AlawDataEnable Indicates whether the codec can be selected for data
epSpecificCodecG711AlawDataEnable transmission.
Data Priority defaultCodecG711AlawDataPriority Priority of this data codec versus the other data codecs.
epSpecificCodecG711AlawDataPriority

Minimum Packetization defaultCodecG711AlawMinPTime Lower boundary for the packetization period.

Time epSpecificCodecG711AlawMinPTime
Maximum defaultCodecG711AlawMaxPTime Upper boundary for the packetization period.

Packetization Time epSpecificCodecG711AlawMaxPTime
G.711 u-law Section
Use Endpoint Specific epSpecificCodecG711MulawEnableConfig Configuration to use for a specific endpoint.
Voice Transmission defaultCodecG711MulawVoiceEnable Indicates whether the codec can be selected for voice
epSpecificCodecG711MulawVoiceEnable transmission.
Voice Priority defaultCodecG711MulawVoicePriority Priority of this voice codec versus the other voice codecs.
epSpecificCodecG711MulawVoicePriority
Data Transmission defaultCodecG711MulawDataEnable Indicates whether the codec can be selected for data
epSpecificCodecG711MulawDataEnable transmission.
Data Priority defaultCodecG711MulawDataPriority Priority of this data codec versus the other data codecs.
epSpecificCodecG711MulawDataPriority
Minimum Packetization defaultCodecG711MulawMinPTime Lower boundary for the packetization period.

Time epSpecificCodecG711MulawMinPTime
Maximum defaultCodecG711MulawMaxPTime Upper boundary for the packetization period.

Packetization Time epSpecificCodecG711MulawMaxPTime
G.723 Section
Use Endpoint Specific epSpecificCodecG723EnableConfig Configuration to use for a specific endpoint.
Voice Transmission defaultCodecG723VoiceEnable Indicates whether the codec can be selected for voice
epSpecificCodecG723VoiceEnable transmission.
Voice Priority defaultCodecG723VoicePriority Priority of this voice codec versus the other voice codecs.
epSpecificCodecG723VoicePriority
Bit Rate defaultCodecG723Bitrate G.723.1 bit rate to use.

epSpecificCodecG723Bitrate
Minimum Packetization defaultCodecG723MinPTime Lower boundary for the packetization period.

Time epSpecificCodecG723MinPTime
Maximum defaultCodecG723MaxPTime Upper boundary for the packetization period.

Packetization Time epSpecificCodecG723MaxPTime
G.726 16Kbps Section
Use Endpoint Specific epSpecificCodecG726r16kbpsEnableConfig Configuration to use for a specific endpoint.
Voice Transmission defaultCodecG726r16kbpsVoiceEnable Indicates whether the codec can be selected for voice
Voice Priority defaultCodecG726r16kbpsVoicePriority Priority of this voice codec versus the other voice codecs.
epSpecificCodecG726r16kbpsVoicePriority
Payload Type defaultCodecG726r16kbpsPayloadType RTP dynamic payload type used in an initial offer.
epSpecificCodecG726r16kbpsPayloadType
Minimum Packetization defaultCodecG726r16kbpsMinPTime Lower boundary for the packetization period.

Time epSpecificCodecG726r16kbpsMinPTime
Maximum defaultCodecG726r16kbpsMaxPTime Upper boundary for the packetization period.

Packetization Time epSpecificCodecG726r16kbpsMaxPTime



Data Transmission defaultCodecG726r32kbpsDataEnable Indicates whether the codec can be selected for data
epSpecificCodecG726r32kbpsDataEnable transmission.
Data Priority defaultCodecG726r32kbpsDataPriority Priority of this data codec versus the other data codecs.
epSpecificCodecG726r32kbpsDataPriority


Data Transmission defaultCodecG726r40kbpsDataEnable Indicates whether the codec can be selected for data
epSpecificCodecG726r40kbpsDataEnable transmission.
Data Priority defaultCodecG726r40kbpsDataPriority Priority of this data codec versus the other data codecs.
epSpecificCodecG726r40kbpsDataPriority



G.729 Section
Use Endpoint Specific epSpecificCodecG729EnableConfig Configuration to use for a specific endpoint.
Voice Transmission defaultCodecG729VoiceEnable Indicates whether the codec can be selected for voice
epSpecificCodecG729VoiceEnable transmission.
Voice Priority defaultCodecG729VoicePriority Priority of this voice codec versus the other voice codecs.
epSpecificCodecG729VoicePriority
Minimum Packetization defaultCodecG729MinPTime Lower boundary for the packetization period.

Time epSpecificCodecG729MinPTime
Maximum defaultCodecG729MaxPTime Upper boundary for the packetization period.

Packetization Time epSpecificCodecG729MaxPTime
Built-In VAD defaultCodecG729VoiceActivityDetection G.729 VAD configuration.

epSpecificCodecG729VoiceActivityDetection
T.38 Section
Use Endpoint Specific epSpecificCodecT38EnableConfig Configuration to use for a specific endpoint.
Enable defaultCodecT38DataEnable If enabled, the T.38 protocol is used for fax transmission.
epSpecificCodecT38DataEnable
Priority defaultCodecT38DataPriority Priority of this data codec versus the other data codecs.
epSpecificCodecT38DataPriority
Redundancy Level defaultCodecT38RedundancyLevel Number of redundancy packets.

epSpecificCodecT38RedundancyLevel
Detection Threshold defaultCodecT38DetectionThreshold Sets the T.38 input signal detection threshold.
epSpecificCodecT38DetectionThreshold
Frame Redundancy defaultCodecT38FinalFramesRedundancy Defines the number of times T.38 packets will be
Level retransmitted.
No Signal defaultCodecT38NoSignalEnable Enables/disables the sending of T.38 no-signal packets.
No Signal Timeout defaultCodecT38NoSignalTimeout The period, in seconds, at which no-signal packets are sent
during a T.38 transmission, in the absence of valid data.
Clear Mode Section
Use Endpoint Specific epSpecificCodecClearModeEnableConfig Configuration to use for a specific endpoint.
Voice Transmission defaultCodecClearModeVoiceEnable Indicates whether the codec can be selected for voice
epSpecificCodecClearModeVoiceEnable transmission.
Voice Priority defaultCodecClearModeVoicePriority Priority of this voice codec versus the other voice codecs.
epSpecificCodecClearModeVoicePriority
Data Transmission defaultCodecClearModeDataEnable Indicates whether the codec can be selected for data
epSpecificCodecClearModeDataEnable transmission.
Data Priority defaultCodecClearModeDataPriority Priority of this data codec versus the other data codecs.
epSpecificCodecClearModeDataPriority
Payload Type defaultCodecClearModePayloadType RTP dynamic payload type used in an initial offer.
epSpecificCodecClearModePayloadType
Minimum Packetization defaultCodecClearModeMinPTime Lower boundary for the packetization period.

Time epSpecificCodecClearModeMinPTime
Maximum defaultCodecClearModeMaxPTime Upper boundary for the packetization period.

Packetization Time epSpecificCodecClearModeMaxPTime

Clear Channel Section
Use Endpoint Specific epSpecificCodecClearChannelEnableConfig Configuration to use for a specific endpoint.
Voice Transmission defaultCodecClearChannelVoiceEnable Indicates whether the codec can be selected for voice
epSpecificCodecClearChannelVoiceEnable transmission.
Voice Priority defaultCodecClearChannelVoicePriority Priority of this voice codec versus the other voice codecs.
epSpecificCodecClearChannelVoicePriority
Data Transmission defaultCodecClearChannelDataEnable Indicates whether the codec can be selected for data
epSpecificCodecClearChannelDataEnable transmission.
Data Priority defaultCodecClearChannelDataPriority Priority of this data codec versus the other data codecs.
epSpecificCodecClearChannelDataPriority
Payload Type defaultCodecClearChannelPayloadType RTP dynamic payload type used in an initial offer.
epSpecificCodecClearChannelPayloadType
Minimum Packetization defaultCodecClearChannelMinPTime Lower boundary for the packetization period.

Time epSpecificCodecClearChannelMinPTime
Maximum defaultCodecClearChannelMaxPTime Upper boundary for the packetization period.

Packetization Time epSpecificCodecClearChannelMaxPTime
X CCD Section
Endpoint Specific epSpecificCodecXCCDEnableConfig Configuration to use for a specific endpoint.
Voice Transmission defaultCodecXCCDVoiceEnable Indicates whether the codec can be selected for voice
epSpecificCodecXCCDVoiceEnable transmission.
Voice Priority defaultCodecXCCDVoicePriority Priority of this voice codec versus the other voice codecs.
epSpecificCodecXCCDVoicePriority
Data Transmission defaultCodecXCCDDataEnable Indicates whether the codec can be selected for data
epSpecificCodecXCCDDataEnable transmission.
Data Priority defaultCodecXCCDDataPriority Priority of this data codec versus the other data codecs.
epSpecificCodecXCCDDataPriority
Payload Type defaultCodecXCCDPayloadType RTP dynamic payload type used in an initial offer.
epSpecificCodecXCCDPayloadType
Minimum Packetization defaultCodecXCCDMinPTime Lower boundary for the packetization period.

Time epSpecificCodecXCCDMinPTime
Maximum defaultCodecXCCDMaxPTime Upper boundary for the packetization period.

Packetization Time epSpecificCodecXccdaxPTime
Security Sub-Page
Security Section
RTP
Mode defaultSecurityRtpMode Defines the RTP payload mode (secure or not secure).
epSpecificSecurityRtpMode
Key Management defaultSecurityKeyManagement Defines the key management protocol for SRTP.
epSpecificSecurityKeyManagement
Encryption defaultSecurityRtpEncryption Defines the encryption type to be used with SRTP.

epSpecificSecurityRtpEncryption
T.38

Allow unsecure T.38 allowUnsecureT38WithSrtp Enables T38 even if the call has been established previously
with secure RTP in SRTP.
RTP Stats
Collection Period Statistics
Period Beginning lastPeriodsStatsPeriodBeginning Date and time of the collection period beginning.
Period End lastPeriodsStatsPeriodEnd Date and time of the collection period end.
Octets Tx lastPeriodsStatsOctetsTransmitted Number of octets transmitted during the collection period.
Octets Rx lastPeriodsStatsOctetsReceived Number of octets received during the collection period.
Packets Tx lastPeriodsStatsPacketsTransmitted Number of packets transmitted during the collection period.
Packets Rx lastPeriodsStatsPacketsReceived Number of packets received during the collection period.
Packets Lost lastPeriodsStatsPacketsLost Number of packets lost during the collection period.
Min. Jitter lastPeriodsStatsMinimumInterarrivalJitter Minimum interarrival time, in milliseconds, during the

collection period.
Max. Jitter lastPeriodsStatsMaximumInterarrivalJitter Maximum interarrival time, in milliseconds, during the

collection period.
Avg. Jitter lastPeriodsStatsAverageInterarrivalJitter Average interarrival time, in milliseconds, during the

collection period.
Min. Latency lastPeriodsStatsMinimumLatency Minimum latency, in milliseconds, during the collection

period.
Max. Latency lastPeriodsStatsMaximumLatency Maximum latency, in milliseconds, during the collection

period.
Avg. Latency lastPeriodsStatsAverageLatency Average latency, in milliseconds, during the collection

period.
Connection Statistics
Octets Tx lastConnectionsStatsOctetsTransmitted Number of octets transmitted during the connection.
Octets Rx lastConnectionsStatsOctetsReceived Number of octets received during the connection.
Packets Tx lastConnectionsStatsPacketsTransmitted Number of packets transmitted during the connection.
Packets Rx lastConnectionsStatsPacketsReceived Number of packets received during the connection.
Packets Lost lastConnectionsStatsPacketsLost Number of packets lost during the connection.
Min. Jitter lastConnectionsStatsMinimumInterarrivalJitter Minimum interarrival time, in milliseconds, during the

connection.
Max. Jitter lastConnectionsStatsMaximumInterarrivalJitter Maximum interarrival time, in milliseconds, during the

connection.
Avg. Jitter lastConnectionsStatsAverageInterarrivalJitter Average interarrival time, in milliseconds, during the

connection.
Min. Latency lastConnectionsStatsMinimumLatency Minimum latency, in milliseconds, during the connection.
Max. Latency lastConnectionsStatsMaximumLatency Maximum latency, in milliseconds, during the connection.
Avg. Latency lastConnectionsStatsAverageLatency Average latency, in milliseconds, during the connection.

Statistics Configuration Section
Collection Period statsCollectionPeriodDuration Specifies the collection period duration (in minutes).
(minutes)
Generate Connection statsPerConnectionNotificationEnable Enables the generation of connection end statistics

End Notification notification.
Generate Collection statsPerPeriodNotificationEnable Enables the generation of period statistics notification.

Period End Notification
Misc Sub-Page
Jitter Buffer Section
Endpoint Specific epSpecificJitterBufferEnableConfig Configuration to use for a specific endpoint.
Level defaultJitterBufferLevel Jitter bufffer level.

epSpecificJitterBufferLevel
Voice Call defaultJitterBufferCustomMinLength Jitter buffer minimum length.

Minimum epSpecificJitterBufferCustomMinLength
Voice Call defaultJitterBufferCustomMaxLength Jitter buffer maximum length.

Maximum epSpecificJitterBufferCustomMaxLength
Data Call defaultVbdJitterBufferType Algorithm to use for managing the jitter buffer during a call.
Playout Type epSpecificCustomVbdJitterBufferType
Data Call defaultVbdJitterBufferCustomMinLength The delay the jitter buffer tries to maintain.
Minimum epSpecificCustomVbdMinLength
Data Call defaultVbdJitterBufferCustomNomLength The delay the jitter buffer uses when a call begins.
Nominal epSpecificCustomVbdNomLength
Data Call defaultVbdJitterBufferCustomMaxLength The highest delay the jitter buffer is allowed to introduce.
Maximum epSpecificCustomVbdMaxLength
DTMF Transport Section
Endpoint Specific epSpecificDtmfTransportEnableConfig Configuration to use for a specific endpoint.
Transport Method defaultDtmfTransportMethod Type of DTMF transport.

epSpecificDtmfTransportMethod
SIP Transport Method interopDtmfTransportMethod Defines the method used to transport DTMFs out-of-band
over the SIP protocol
Payload Type defaultDtmfTransportPayloadType RTP dynamic payload type used for telephone-event in an
epSpecificDtmfTransportPayloadType initial offer.
Machine Detection Section
Endpoint Specific specificMachineDetectionEnableConfig Configuration to use for a specific interface.
CNG Tone Detection defaultMachineDetectionCngToneDetection Enables fax calling tone (CNG tone) detection.
specificMachineDetectionCngToneDetection
CED Tone Detection defaultMachineDetectionCedToneDetection Enables CED tone detection.

specificMachineDetectionCedToneDetection

Telephony Page Software Configuration Guide
V.21 Modulation defaultMachineDetectionV21ModulationDetecti Enables fax V.21 modulation detection.

Detection on
specificMachineDetectionV21ModulationDetecti
on
Behavior on CED Tone defaultMachineDetectionBehaviorOnCedToneD Defines the behavior of the unit upon detection of a CED
Detection etection tone.
specificMachineDetectionBehaviorOnCedTone
Detection
Base Ports Section
RTP ipTransportRtpBasePort UDP base port for the RTP/RTCP protocols.
SRTP ipTransportSrtpBasePort UDP base port for the SRTP/SRTCP protocols.
T.38 ipTransportT38BasePort T.38 base port.
Telephony Page
DTMF Maps Sub-Page
First DTMF Timeout dtmfMapTimeoutFirstDtmf Time the user has to enter the first DTMF after the dial tone.
Inter DTMF Timeout dtmfMapTimeoutInterDtmf Value of the “T” DTMF in the DTMF map strings.
Completion Timeout dtmfMapTimeoutCompletion Total time the user has to dial the DTMF sequence.
DTMF Maps Digit DtmfMapDigitDetection Determines when a digit is processed through the DTMF
Detection (FXO/FXS) maps.
Allowed DTMF Map Section
Index callDtmfMapAllowedIndex Accepted DTMF map index for this row.
Enable callDtmfMapAllowedEnable Enables/Disables the row.
Apply to callDtmfMapAllowedApplyTo Entity to which apply the DTMF map.
Endpoint callDtmfMapAllowedEpId String that identifies an endpoint in other tables.
DTMF Map callDtmfMapAllowedDtmfMap DTMF map that is considered valid when dialed.
Transformation callDtmfMapAllowedDtmfTransformation Transformation to apply to the signalled DTMF before using

it as call destination.
Target callDtmfMapAllowedTargetHost Target to use when the DTMF map matches.
Emergency callDtmfMapAllowedEmergency Enables/Disables the emergency process of the call.
Refused DTMF Map Section
Index callDtmfMapRefuseIndex Refused DTMF map index for this row.

Appendix D - Web Interface – SNMP Variables Mapping Telephony Page
Enable callDtmfMapRefuseEnable If enabled, this DTMF map is recognised and refused only if
it is also valid.
Apply to callDtmfMapRefuseApplyTo Sets the entity to which apply the DTMF map.
Endpoint callDtmfMapRefuseEpId String that identifies an endpoint in other tables.
DTMF Map callDtmfMapRefuseDtmfMap DTMF map that is considered invalid when dialed.
DTMF Map Timeout Section
Endpoint EpSpecificDtmfMapTimeoutEpId String that identifies an endpoint in other tables.
Override EpSpecificDtmfMapTimeoutEnableConfig Defines the configuration to use for a specific endpoint.
First DTMF Timeout EpSpecificDtmfMapTimeoutFirstDtmf Time the user has to enter the first DTMF after the dial tone.
Inter DTMF Timeout EpSpecificDtmfMapTimeoutInterDtmf Value of the 'T' DTMF in the DTMF map strings.
Completion Timeout EpSpecificDtmfMapTimeoutCompletion Total time the user has to dial the DTMF sequence.
Call Forward Sub-Page

Call Forward On Busy Section
Endpoint Specific epSpecificForwardOnBusyEnableConfig Defines the configuration to use for a specific endpoint.
Allow Activation via defaultForwardOnBusyEnable Enables/Disables the call forward on busy service.
Handset epSpecificForwardOnBusyEnable
DTMF Map Activation defaultForwardOnBusyDtmfMapActivation DTMF map the user can dial to enable the application of the
service.
DTMF Map defaultForwardOnBusyDtmfMapDeactivation DTMF map the user can dial to disable the application of the
Deactivation service.
Activation forwardOnBusyConfigActivation Activation status of the call forward on busy service.
Forwarding Address: forwardOnBusyConfigForwardingAddress Address or telephone number to which the user wants to
forward calls.
Call Forward On No Answer Section
Endpoint Specific epSpecificForwardNoAnswerEnableConfig Defines the configuration to use for a specific endpoint.
Allow Activation via defaultForwardNoAnswerEnable Enables/Disables the call forward on no answer service.
Handset epSpecificForwardNoAnswerEnable
DTMF Map Activation defaultForwardNoAnswerDtmfMapActivation DTMF map the user can dial to enable the application of the
service.
DTMF Map defaultForwardNoAnswerDtmfMapDeactivation DTMF map the user can dial to disable the application of the
Deactivation service.
Timeout defaultForwardNoAnswerTimeout Time, in milliseconds, the telephone keeps ringing before

epSpecificForwardNoAnswerTimeout the call forwarding activates.
Activation forwardNoAnswerConfigActivation Activation status of the call forward on no answer service.
Forwarding Address: forwardNoAnswerConfigForwardingAddress Address or telephone number to which the user wants to
forward calls.

Call Forward Unconditional Section
Endpoint Specific epSpecificForwardUnconditionalEnableConfig Defines the configuration to use for a specific endpoint.
Allow Activation via defaultForwardUnconditionalEnable Enables/Disables the unconditional call forward service.
Handset epSpecificForwardUnconditionalEnable
DTMF Map Activation defaultForwardUnconditionalDtmfMapActivation DTMF map the user can dial to enable the application of the
service.
DTMF Map defaultForwardUnconditionalDtmfMapDeactivati DTMF map the user can dial to disable the application of the
Deactivation on service.
Activation forwardUnconditionalConfigActivation Activation state of the unconditional call forward service.
Forwarding Address: forwardUnconditionalConfigForwardingAddress Address or telephone number to which the user wants to
forward calls.
Services Sub-Page
Service Section
Blind Transfer transferStatusBlindState Status of the blind transfer service.
Attended Transfer transferStatusAttendedState Status of the attended transfer service.
Call Waiting callWaitingStatusState Status of the call waiting service.
Conference conferenceStatusState Status of the conference service.
Hold holdStatusState Status of the holding service.
Second Call secondCallStatusState Status of the second call service.
Active Call Completion Section
Endpoint callCompletionConfigEpId Identification of the endpoint using this call completion

service.
Type callCompletionConfigType The type of the call completion.
Target Address callCompletionConfigTargetAddress The target address of the call completion.
Target State callCompletionConfigTargetState The state of the call completion target.
Services Configuration Section
Endpoint Specific epSpecificCallEnableConfig Defines the configuration to use for a specific endpoint.
Hook Flash Processing defaultCallHookFlashProcessing Selects how to process hook-flash detection.

epSpecificCallHookFlashProcessing
Automatic Call
Endpoint Specific epSpecificAutoCallEnableConfig Defines the configuration to use for a specific endpoint.
Automatic Call defaultAutoCallEnable Enables/Disables the automatic call service. This service
Activation epSpecificAutoCallEnable provides a 'redphone'-like experience.

Automatic Call Target defaultAutoCallTargetAddress Address or telephone number that the user wants to
epSpecificAutoCallTargetAddress automatically call.
Call Completion
Endpoint Specific epSpecificCallCompletionEnableConfig Defines the configuration to use for a specific endpoint.
Allow CCBS Activation defaultCallCompletionBusySubscriberEnable Enables/Disables the call completion busy subscriber
Via Handset epSpecificCallCompletionBusySubscriberEnabl (CCBS) service.
e
Allow CCNR Activation defaultCallCompletionNoResponseEnable Enables/Disables the call completion no response (CCNR)
Via Handset epSpecificCallCompletionNoResponseEnable service.
CCBS DTMF Map defaultCallCompletionBusySubscriberDtmfMap DTMF map the user can dial to enable the application of the
Activation Activation call completion busy subscriber (CCBS) service.
CCNR DTMF Map defaultCallCompletionNoResponseDtmfMapAct DTMF map the user can dial to enable the application of the
Activation ivation call completion no response (CCNR) service.
DTMF Map defaultCallCompletionDtmfMapDeactivation DTMF map the user can dial to disable the application of the
Deactivation call completion busy subscriber (CCBS) and call completion
no response (CCNR) services.
Expiration Timeout defaultCallCompletionExpirationTimeout Defines the delay after the call completion activation to
automatically deactivate the call completion if the call is not
completed.
Method defaultCallCompletionMethod Selects the call completion method to detect that the call
completion destination is ready to complete the call.
Auto Reactivate defaultCallCompletionAutoReactivateEnable Enables/Disables the call compleion auto reactivation.
Auto Reactivate Delay defaultCallCompletionAutoReactivateDelay Defines the minimal delay to wait before executing a call
completion after its activation.
Early-Media Behaviour defaultCallCompletionEarlyMediaBehaviour Defines how the call completion service needs to interpret
the reception of a progress message with early media.
Polling Interval defaultCallCompletionPollingInterval Defines the delay between the calls to the call completion
target used for the polling mechanism.
Call Transfer
Endpoint Specific epSpecificTransferEnableConfig Defines the configuration to use for a specific endpoint.
Blind Transfer defaultTransferBlindEnable Enables/Disables the blind call transfer service.

Activation epSpecificTransferBlindEnable
Attended Transfer defaultTransferAttendedEnable Enables/Disables the attended call transfer service.

Activation epSpecificTransferAttendedEnable
Call Waiting
Endpoint Specific epSpecificCallWaitingEnableConfig Defines the configuration to use for a specific endpoint.
Call Waiting Activation defaultCallWaitingEnable Enables/Disables the call waiting service.

epSpecificCallWaitingEnable
Cancel DTMF Map defaultCallWaitingCancelDtmfMap Default DTMF Map to Cancel the Call Waiting Service
Conference
Endpoint Specific epSpecificConferenceEnableConfig Defines the configuration to use for a specific endpoint.
Conference Activation defaultConferenceEnable Enables/Disables the call conference service.

epSpecificConferenceEnable
Delayed Hotline
Endpoint Specific epSpecificDelayedHotlineEnableConfig Defines the configuration to use for a specific endpoint.
Delayed Hotline defaultDelayedHotlineEnable Enables/Disables the delayed hotline service.

Activation epSpecificDelayedHotlineEnable
Delayed Hotline defaultDelayedHotlineCondition Selects the condition(s) that activate the delayed hotline.
Condition epSpecificDelayedHotlineCondition
Delayed Hotline Target defaultDelayedHotlineTargetAddress Address or telephone number of the target of the delayed
epSpecificDelayedHotlineTargetAddress hotline.
Direct IP Address Call

Direct IP Address Call defaultCallAllowDirectIp Enables/Disables the direct IP address call service.
Activation
Hold
Endpoint Specific epSpecificHoldEnableConfig Defines the configuration to use for a specific endpoint.
Hold Activation defaultHoldEnable Enables/Disables the holding service.

epSpecificHoldEnable
Second Call
Endpoint Specific epSpecificSecondCallEnableConfig Defines the configuration to use for a specific endpoint.
Second Call Activation defaultSecondCallEnable Enables/Disables the second call service.

epSpecificSecondCallEnable
Tone Customization Sub-Page

Select Tone countryCustomizationToneTone Tone to customize.
Override Current Tone countryCustomizationToneOverride Allows overriding the default country tone setting.
Values
Current Tone Definition section
Frequencies
Value
countryToneStatusPattern Pattern description of the currently used tone for the country
Power
Loop Count
Current Tone States section
State
On/Off
Frequencies
countryToneStatusPattern Pattern description of the currently used tone for the country
Duration
Loop
Next State
Overriden Tone Definition section
Frequencies
Value
countryCustomizationTonePattern Pattern description of the custom tone.
Power
Loop Count

Overriden Tone States section
State
On/Off
Frequencies
countryCustomizationTonePattern Pattern description of the custom tone.
Duration
Loop
Next State
Music on Hold Sub-Page

Transfer (command) Saves the settings and transfers the MP3 file now.
CancelTransfer (command) Saves the settings and stops a file transfer in progress.
Status Section
File Status fileStatus Status of the MP3 file in the unit.
Last Transfer Result lastTransferStatus Status of the last file transfer attempt.
Last Successful lastTransferDateTime Date and time of the last successful music file transfer.
Transfer
Music On Hold Configuration Section
Streaming musicOnHoldStreamingEnable Indicate whether or not the unit should play music when
being put on hold.
Transfer Configuration Section
URL fileUrl URL to a MP3 file which will be loaded at unit startup and
reloaded every time the ReloadInterval elapsed.
User Name username When authentication is required by the remote file server,
this variable will be used as the username.
Password password When authentication is required by the remote file server,

this variable will be used as the password.
Reload Interval reloadInterval Time, in hours, between attempts to load the MP3 file.
Misc Sub-Page
Country Section
Country Selection countrySelection List of predefined country settings.

Call Router Page Software Configuration Guide
Custom Tone Section
Override countryCustomizationToneOverride Overrides the default country tone setting.
Pattern countryCustomizationTonePattern Pattern description of the custom tone.
Call Detail Record Section
Syslog Remote Host syslogRemoteHost Host name and port number of the device that archives CDR
log entries.
Syslog Format syslogFormat Specifies the format of the syslog Call Detail Record.
Syslog Facility syslogFacility Syslog facility used by the unit to route the Call Detail
Record messages.
Call Router Page
Status Sub-Page
Config Modified configModifiedStatus Shows whether the configuration of the call routing was
Route Section
Type routeStatusType Displays the associated route type.
Source routeStatusSourceCriteria Source criteria to match to apply the route.
Properties Criteria routeStatusPropertiesCriteria Call properties criteria to match to apply the route.
Expression Criteria routeStatusExpressionCriteria Expression criteria to match to apply the route.
Mappings routeStatusMapping Name of the properties manipulation to apply to the call if

the criteria match.
Signaling Properties routeStatusSignalingProperties Name of the signaling properties to apply to the call.
Destination routeStatusDestination Destination to apply to the call if it matches the criteria.
Signaling Properties Section
Name signalingPropertiesStatusName Name of the Signaling properties defined by this row.
Early Connect signalingPropertiesStatusEarlyConnect Enables/Disables the early connect feature.
Early Disconnect signalingPropertiesStatusEarlyDisconnect Enables/Disables the early disconnect feature.
Destination Host signalingPropertiesStatusDestinationHost SIP messages destination.

Appendix D - Web Interface – SNMP Variables Mapping Call Router Page
Mapping Section
Criteria mappingTypeStatusCriteria Expression or call property to compare with the call and
mappingExpressionStatusCriteria match in order to apply the properties manipulation.
Transformation mappingTypeStatusTransformation Call properties to transform and transformation to apply to

mappingExpressionStatusTransformation the call properties.
Sub Mapping mappingExpressionStatusSubMappings Name of a subsequent properties manipulation to execute.
Hunt Section
Name huntStatusName Name of the hunt defined by this row.
Destinations huntStatusDestinations List of hunt destinations separated by comma.
Selection Algorithm huntStatusSelectionAlgorithm Destination selection algorithm.
Timeout huntStatusTimeout Maximal time allowed to the destination to handle the call.
Causes huntStatusCauses List of call rejection causes to continue the hunt.
SIP Redirects Section
Name sipRedirectStatusName Name of the SIP Redirect defined by this row.
Destination Host sipRedirectStatusDestination Host address inserted in the Moved Temporarily response.
Available Interfaces Section
Index interfaceStatusIndex Unique identifier of the row in the table.
Name interfaceStatusName Name of the insterface.
Route Configuration Sub-Page

ApplyConfig (command) Applies the call routing configuration.
RollbackConfig (command) Rolls back the current configuration to the running

configuration as showed in the status. The current
configuration will be lost.
Route Section
routeUp Moves the current row upside.
routeDown Moves the current row downside.
routeInsert Inserts a new row before this row.

routeInsertRoute Inserts a new row at the end of the Route table.
routeDelete Deletes this row.

Configure Route Panel
Source routeSourceCriteria Source criteria to match to apply the route.
Properties Criteria routePropertiesCriteria Call properties criteria to match to apply the route.
Expression Criteria routeExpressionCriteria Expression criteria to match to apply the route.
Mappings routeMapping Name of the properties manipulation to apply to the call if

the criteria match.
Signaling Properties routeSignalingProperties Name of the signaling properties to apply to the call.
Destination routeDestination Destination to apply to the call if the call matches the criteria.
Config Status routeConfigStatus Configuration status of the row.
Mapping Type Section
mappingTypeUp Moves the current row upside.
mappingTypeDown Moves the current row downside.
mappingTypeInsert Inserts a new row before this row.

mappingTypeInsertMappingType Inserts a new row at the end of the MappingType table.
mappingTypeDelete Deletes this row.
Configure Mapping Type Panel
Name mappingTypeName Name of the the properties manipulation.
Criteria mappingTypeCriteria Call properties that the service must compare with the call
and match in order to apply the properties manipulation.
Transformation mappingTypeTransformation Call properties to transform.
Config Status mappingTypeConfigStatus It indicates whether the configuration of the row is valid.
Mapping Expression Section
mappingExpressionUp Moves the current row upside.
mappingExpressionDown Moves the current row downside.
mappingExpressionInsert Inserts a new row before this row.

mappingExpressionInsertMappingExpression Inserts a new row at the end of the MappingExpression
table.
mappingExpressionDelete Deletes this row.
Configure Mapping Expression Panel
Name mappingExpressionName Name of the properties manipulation.
Criteria mappingExpressionExpressionCriteria Expression to compare with the call and match in order to
apply the properties manipulation.
Transformation mappingExpressionTransformation Transformation to apply to the call properties.

Appendix D - Web Interface – SNMP Variables Mapping Call Router Page
Sub Mappings mappingExpressionSubMapping Name of a subsequent properties manipulation to execute.
Config Status mappingExpressionConfigStatus It indicates whether the configuration of the row is valid.
Signaling Properties Section
signalingPropertiesUp Moves the current row upside.
signalingPropertiesDown Moves the current row downside.
signalingPropertiesInsert Inserts a new row before this row.

signalingPropertiesInsertSignalingProperties Inserts a new row at the end of the Signaling Properties
table.
signalingPropertiesDelete Deletes this row.
Configure Signaling Properties Panel
Name signalingPropertiesName Name of the Signaling properties defined by this row.
Early Connect signalingPropertiesEarlyConnect Enables/Disables the early connect feature.
Early Disconnect signalingPropertiesEarlyDisconnect Enables/Disables the early disconnect feature.
Destination Host signalingPropertiesDestinationHost SIP messages destination.
Allow 180 with SDP signalingPropertiesAllow180Sdp Enables/Disables the 180 with SDP allowed.
Allow 183 without SDP signalingPropertiesAllow183NoSdp Enables/Disables the 183 without SDP allowed.
Privacy signalingPropertiesPrivacy Sets the privacy level of the call.
SIP Headers signalingPropertiesSipHeadersTranslation Name of the SIP headers translation to apply to the call.
Translations
Call Properties signalingPropertiesCallPropertiesTranslation Name of the call properties translation to apply to the call.
Translations
Config Status signalingPropertiesConfigStatus Configuration status of the row.
SIP Headers Translations Section
sipHeadersTranslationUp Moves the current row upside.
sipHeadersTranslationDown Moves the current row downside.
sipHeadersTranslationInsert Inserts a new row before this row.

sipHeadersTranslation Inserts a new row at the end of the SIP Headers Translation
table.
sipHeadersTranslationDelete Deletes this row.
Configure SIP Headers Translation Panel
Name sipHeadersTranslationName Name of the SIP headers translation defined by this row.
SIP Header sipHeadersTranslationSipHeader Sets which SIP header is modified by this translation.
Built From sipHeadersTranslationBuiltFrom Sets what information is used to build the selected SIP
header.

Fix Value sipHeadersTranslationFixValue Fix value to be inserted in the SIP header.
Config Status sipHeadersTranslationConfigStatus Configuration status of the row.
Call Properties Translations Section
callPropertiesTranslationUp Moves the current row upside.
callPropertiesTranslationDown Moves the current row downside.
callPropertiesTranslationInsert Inserts a new row before this row.

callPropertiesTranslationInsertCallPropertiesTra Inserts a new row at the end of the SIP Headers Translation
nslation table.
callPropertiesTranslationDelete Deletes this row.
Configure Call Properties Translations Panel
Name callPropertiesTranslationName Name of the call properties translation defined by this row.
Call Property callPropertiesTranslationCallProperty Sets which call property is modified by this translation.
Built From callPropertiesTranslationBuiltFrom Sets what information is used to build the selected call
property.
Fix Value callPropertiesTranslationFixValue Fix value to be inserted in the call property.
Config Status callPropertiesTranslationConfigStatus Configuration status of the row.
Hunt Section
huntUp Moves the current row upside.
huntDown Moves the current row downside.
huntInsert Inserts a new row before this row.

huntInsertHunt Inserts a new row at the end of the Hunt table.
huntDelete Deletes this row.
Configure Hunt Panel
Name huntName Name of the hunt defined by this row.
Destinations huntDestinations List of hunt destinations separated by comma.
Selection Algorithm huntSelectionAlgorithm Destination selection algorithm.
Timeout huntTimeout Maximal time allowed to the destination to handle the call.
Causes huntCauses List of call rejection causes to continue the hunt.
Config Status huntConfigStatus Configuration status of the row.

Appendix D - Web Interface – SNMP Variables Mapping Management Page
SIP Redirects Section
sipRedirectUp Moves the current row upside.
sipRedirectDown Moves the current row downside.
sipRedirectInsert Inserts a new row before this row.

sipRedirectInsertSipRedirect Inserts a new row at the end of the SIP Redirects table.
sipRedirectDelete Deletes this row.
Configure SIP Redirects Panel
Name sipRedirectName Name of the SIP Redirect defined by this row.
Destination Host sipRedirectDestinations Host address to be inserted in the Moved Temporarily

response.
Config Status sipRedirectConfigStatus Configuration status of the row. It indicates whether the
configuration of the row is valid.
Auto-Routing Sub-Page
Auto-routing autoRoutingEnable Enables/Disables the automatic insertion of default routes

for selected endpoints.
Criteria Type autoRoutingCriteriaType Determines the type of criteria to use to create automatic
rule from SIP to the telephony endpoints.
Incoming Mappings autoRoutingIncomingMappings Name of the properties manipulations associated with the
route from the SIP gateway to the endpoint.
Outgoing Mappings autoRoutingOutgoingMappings Name of the properties manipulations associated with the
route from the endpoint to the SIP gateway.
Incoming Signaling autoRoutingIncomingSignalingProperties Name of the signaling properties associated with the route
Properties from the SIP gateway to the endpoint.
Outgoing Signaling autoRoutingOutgoingSignalingProperties Name of the signaling properties associated with the route
Properties from the endpoint to the SIP gateway.
Endpoints auto-routing section
Endpoint autoRoutingEpId Character string that identifies an endpoint in other tables.
Auto-routable autoRoutingAutoroutable Determines whether or not automatic routes are generated

for the endpoint when auto-routing is enabled.
Auto-routing Gateway autoRoutingAutoRoutingGateway Name of the SIP gateway to use as the destination of
outgoing calls and the source of incoming calls when
generating auto-routing rules.
E164 autoRoutingE164 The telephone number associated with this endpoint, if any.
SIP Username autoRoutingSipUsername The SIP username associated with this endpoint, if any.
Name autoRoutingName The FriendlyName associated with this endpoint, if any.
Management Page

Management Page Software Configuration Guide
Configuration Scripts Sub-Page

ConfiguredScriptExport (command) Command to export the configuration script.
ConfiguredScriptsTransferAndRun (command) Command to launch the configuration scripts download.
Scripts Status Section
Current State (Export) scriptsStatsCurrentTransferState The current state of the configuration script transfer and
execution.
Current State ScriptsStatsCurrentExportState The current state of the configuration script exportation.
(Execute)
Last Result (Export) scriptsStatsLastTransferResult Result of the last configuration scripts transfer command.
Last Result (Execute) ScriptsStatsLastExportResult Result of the last configuration script exportation command.
Last Successful scriptsStatsLastTransferDateTime Date and time of the last successful configuration script
(Export) transfer command.
Last Successful ScriptsStatsLastExportDateTime Date and time of the last successful configuration script
(Execute) exportation and transfer command since the last reset to
default settings.
Export Script Section
Content scriptExportContent Content to export in the generated configuration script.
Service Name ScriptExportServiceName Name of the service from which to export configuration.
Send To URL scriptExportUrl URL where to send the configuration script exported.
Privacy Key scriptExportSecretKey Key used to encrypt the configuration script to export.
Execute Scripts Section
Generic File Name scriptGenericFileName Name of the generic configuration script.
Specific File Name scriptSpecificFileName Name of the specific configuration script.
Transfer Protocol scriptsTransferProtocol Protocol used to transfer the configuration script files.
Host Name scriptsTransferSrvHostname Configuration scripts server hostname and port.
Location scriptsLocation Path to the location of the configuration scripts.
User Name scriptsTransferUsername User name used to transfer the configuration script.
Password scriptsTransferPassword Password used to transfer the configuration script.
Privacy Key scriptsSecretKey Key used to decrypt encrypted configuration scripts.
Allow Repeated scriptsAllowRepeatedExecution Allows the execution of a script even if it is identical to the
Execution last executed script.
Automatically Update Scripts Section
Update On Restart scriptsTransferOnRestartEnable Enables automatic configuration scripts transfer on restart.
Update Periodically scriptsTransferPeriodicEnable Enables automatic periodic configuration scripts transfer.

Time Unit scriptsTransferPeriodicTimeUnit Time unit for the variable scriptsTransferInterval.
Period scriptsTransferInterval Time interval between automatic configuration scripts

transfer.
Time Of Day scriptsTransferTimeOfDay Time when the automatic configuration scripts transfer
occurs.
DHCP Download scriptsDhcpDownloadEnable DHCP Triggered Script Support.

Enable
Transfer Scripts Through Web Browser Section
Upload Parameters N/A N/A
Content scriptExportContent Content to export in the generated configuration script.
Privacy Key scriptsSecretKey Key used to decrypt encrypted configuration scripts.
Backup / Restore Sub-Page

ConfiguredBackupImage (command) Command to launch the configuration backup.
ConfiguredRestoreImage (command) Command to launch the configuration restore.
Status Section
Last Backup Result imageBackupStatus Result of the last configuration backup command.
Last Restore Result imageRestoreStatus Result of the last configuration restore command.
Image Configuration Section
File Name imageFileName Name of the file used to backup (save) and restore (load)
the unit’s configuration.
Transfer Protocol imageTransferProtocol Protocol used to upload a configuration image during

backup and transfer during restore.
Host Name imageTransferSrvHostname Configuration backup/restore server hostname and port.
Location imageLocation Path to the location of the configuration image file.
User Name imageTransferUsername User name used to transfer the configuration image.
Password imageTransferPassword Password used to transfer the configuration image.
Content imageBackupContent Defines what to include in the backup.
Privacy Algorithm imagePrivacyAlgo Enables/disables decryption of the configuration image.
Privacy Key imageSecretKey Key used to decrypt or encrypt a configuration image.
Firmware Upgrade Sub-Page

Install (command) Command to launch the firmware download.

Status Section
Firmware Pack status Indicates the current status of the Firmware Pack Updater.
Updater Status
Last Installation Result mfpLastInstallationResult Result of the last install command.
Last Successful mfpLastInstallationDateTime Date and time of the last successful install command.
Installation
Firmware Packs Installed
Name mfpSelectionMfpname Name of the Firmware Pack to install.
Version mfpVersion Version of the MFP to install.
Profile mfpProfileName Name of the profile.
Bank mfpInstalledInfoMfpBank Bank where the MFP is installed.
Rollback (command) Launches the rollback of the previously installed MFP found
in recovery bank.
Firmware Packs Configuration
Language languageSelection Language.
Version mfpVersion Version of the MFP to install.
Automatic Restart automaticRestartEnable Enables the firmware pack updater to automatically restart
Enable the system when needed for completing a firmware update
operation.
Automatic Restart automaticRestartGraceDelay Configures the grace delay in minutes that the unit waits for
Grace Delay all telephony calls to be terminated before the automatic
restart can occur.
Firmware Pack 1 mfpSelectionMfpName Name of the Firmware Pack to install.
Transfer Configuration
Location mfpLocation Path to the directory containing MFPs.
Transfer Protocol mfpTransferProtocol Protocol to use to access the update tree.
User Name mfpTransferUsername User name to use to access the update tree.
Password mfpTransferPassword Password to use to access the update tree.
Host Name mfpTransferSrvHostname Name or IP address and port of the Update Files server.
Certificates Sub-Page

Host Certificates Section
File Name hostCertificatesInfoFileName Name of the certificate file.
Issued To hostCertificatesInfoIssuedTo Certificate subject name. This is the common name that
must match the host being authenticated.
Issued By hostCertificatesInfoIssuedBy Certificate issuer name. This is the certificate authority that
signed this certificate.
Valid From hostCertificatesInfoValidFrom Certificate lower bound validity duration range.
Valid To hostCertificatesInfoValidTo Certificate higher bound validity duration range.
Usage hostCertificatesAuthenticationUsage Identifies in which role or context a certificate can be used

by the host it authenticates.
hostCertificatesInfoDelete Removes the certificate from the unit.
Others Certificates Section
File Name othersCertificatesInfoFileName Name of the certificate file.
Issued To othersCertificatesInfoIssuedTo Certificate subject name. This is the common name that
must match the host being authenticated.
Issued By othersCertificatesInfoIssuedBy Certificate issuer name. This is the certificate authority that
signed this certificate.
Valid From othersCertificatesInfoValidFrom Certificate lower bound validity duration range.
Valid To othersCertificatesInfoValidTo Certificate higher bound validity duration range.
Usage othersCertificatesAuthenticationUsage Identifies in which role or context a certificate can be used

by the host it authenticates.
CA othersCertificatesInfoCertificateAuthority Indicates if the certificate is a CA certificate.
othersCertificatesInfoDelete Removes the certificate from the unit.
Host Certificate Associations Section
File Name hostCertificateAssociationFileName Certificate file name.
SIP hostCertificateAssociationSip Specifies if this certificate can be used for SIP security.
Web hostCertificateAssociationWeb Specifies if this certificate can be used for Web security.
EAP hostCertificateAssociationEap Specifies if this certificate can be used for EAP security.
Certificate Authorities Section
File Name certificateAuthoritiesFileName Certificate authority (CA) file name.
Override OCSP URL certificateAuthoritiesOverrideIssuedCertificateO Defines a specific OCSP URL to use for certificate
cspUrl revocation status of certificates issued by this certificate
authority (CA).

SNMP Configuration Section
SNMP Listening Port port Port on which the SNMP service should listen for incoming
SNMP requests.
Enable SNMP V1 enableSnmpV1 Specifies if a user can connect to the system by using
SNMPv1.
SNMPv2.
SNMPv3.
Authentication Protocol authProtocol Protocol to use with SNMPv3.
Privacy Protocol privProtocol Protocol to use with SNMPv3.
Privacy Password privPassword Password to use with SNMPv3 when using DES privacy.
Community community String to use for the community field of SNMPv1 and
SNMPv2 read-write commands and traps.
Enable SNMP Trap enableTrap Specifies if traps can be sent.
Trap Destination trapDest Addresses/FQDNs and ports where to send traps.Up to 5

destinations can be specified by using a comma between
them. The port numbers are optional.
Access Control Sub-Page
Users Section
User Name usersUserName Contains the user name.
Password usersPassword Contains the user's password.
Access Rights usersAccessRights Defines the access rights template applying to a user.
InsertUser Inserts a new user in the Users table.
Delete (Row Command) Deletes this row.
Services Access Control Type Section
Service servicesAaaTypeService Service name for which the Aaa types are configured.
Authentication Type servicesAaaTypeAuthenticationType Authentication type a service uses for incoming

authentication requests.
Accounting Type servicesAaaTypeAccountingType Accounting type a service uses once a user is successfully
authenticated on the unit.
Radius Servers Section
Authentication
Host radiusServersAuthenticationHost Hostname and port of a Radius server used for

authentication requests.
Server Secret radiusServersAuthenticationSecret Secret key shared between the Radius server and the unit.
Accounting

Host radiusServersAccountingHost Hostname and port of a Radius server used for accounting
requests.
Server Secret radiusServersAccountingSecret Secret key shared between the Radius server and the unit.
General
Server Request radiusServersTimeoutS Maximum time, in seconds, the unit waits for a reply from a
Timeout Radius server.
Radius Users Access radiusUserAccessRights Defines the access rights template applying to all Radius
Rights users.
File Sub-Page
Internal Files Section
Name filesFilename Relative path and name of the file.
Description filesFileDescription Textual description describing the content of the file.
Size filesFileSize File size of the associated file.
filesDelete Deletes this row.
Misc Sub-Page
System Management Section
Network Interface managementInterface Specifies to which network interface system management

services are bound.

A P P E N D I X
E Glossary
10 BaseT
An Ethernet local area network that works on twisted pair wiring.
100 BaseT
A newer version of Ethernet that operates at 10 times the speed of a 10 BaseT Ethernet.
Access Device
Device capable of sending or receiving data over a data communications channel.
Accounting
Accounting measures the resources a user consumes during access. This can include the amount of system
time or the amount of data a user has sent and/or received during a session. Accounting is carried out by
logging of session statistics and usage information and is used for authorization control, billing, trend analysis,
resource utilization, and capacity planning activities.
A-Law
The ITU-T companding standard used in the conversion between analog and digital signals in PCM (Pulse
Code Modulation) systems. A-law is used primarily in European telephone networks and contrasts with the
North American mu (µ)-law standard. See also mu (µ)-law.
ANI
In CAS signalling, the sending of the calling numbers is known as Automatic Number Identification.
AOC
In ISDN signalling, an Advice Of Charge (AOC-D) message is sent to advise of the current charge (D)uring a
call or an AOC-E message is sent to advise of the total charge at the (E)nd of a call.
Area Code
The preliminary digits that a user must dial to be connected to a particular outgoing trunk group or line. In North
America, an area code has three digits and is used with a NXX (office code) number. For instance, in the North
American telephone number 561-955-1212, the numbers are defined as follows:
Table 351: North American Numbering Plan
No. Description
561 Area Code, corresponding to a geographical zone in a non-LNP (Local Number Portability)
network.
955 NXX (office code), which corresponds to a specific area such as a city region.
1212 Unique number to reach a specific destination.

Appendix E - Glossary
Outside North America, the area code may have any number of digits, depending on the national
telecommunication regulation of the country. In France, for instance, the numbering terminology is xZABPQ
12 34, where:
Table 352: France Numbering Plan
No. Description
x Operator forwarding the call. This prefix can be made of 4 digits.

Z Geographical (regional) zone of the number (in France, there are five zones). It has two digits.
ABPQ First four digits corresponding to a local zone defined by central offices.
12 34 Unique number to reach a specific destination.
In this context, the area code corresponds to the Z portion of the numbering plan. Because virtually every
country has a different dialing plan nomenclature, it is recommended to identify the equivalent of an area code
for the location of your communication unit.
Authentication
Authentication provides a way of identifying a user, typically by having the user enter a valid user name and
valid password before access is granted. The process of authentication is based on each user having a unique
set of criteria for gaining access. The AAA server compares a user's authentication credentials with other user
credentials stored in a database. If the credentials match, the user is granted access to the network. If the
credentials are at variance, authentication fails and network access is denied.
Basic Rate Interface (BRI)

An Integrated Services Digital Network configuration defined in the physical layer standard I.430 produced by
the ITU. This configuration consists of two 64 kbit/s “bearer” channels (B channels) and one 16 kbit/s “data”
channel (D channel). The B channels are used for voice or user data, and the D channel is used for any
combination of: data, control/signalling and X.25 packet networking. The two B channels can be bonded
together giving a total data rate of 128 kbit/s. BRI is the kind of ISDN interface most likely to be found in
residential service.
Call Routing
Calls through the Aastra can be routed based on a set of routing criteria.
Channel Associated Signalling (CAS)

With this method of signalling, each traffic channel has a dedicated signalling channel. In other words the
signalling for a particular traffic circuit is permanently associated with that circuit. Channel-associated call-
control is still widely used today mostly in South America, Africa, Australia and in Europe.
Country Code (CC)

In international direct telephone dialing, a code that consists of 1-, 2-, or 3-digit numbers in which the first digit
designates the region and succeeding digits, if any, designate the country.
Dialed Number Identification Service (DNIS)

DNIS is a telephone service that identifies for the receiver of a call the number that the caller dialed. It's a
common feature of 800 and 900 lines. If you have multiple 800 or 900 numbers to the same destination, DNIS
tells which number was called. DNIS works by passing the touch tone digits (dual tone multi frequency or MF
digits) to the destination where a special facility can read and display them or make them available for call
center programming.
Digital Subscriber Lines (DSL)

A technology for bringing high-bandwidth information to homes and small businesses over ordinary copper
telephone lines. xDSL refers to different variations of DSL, such as ADSL, HDSL, and RADSL.

Software Configuration Guide
Distinguished Encoding Rules (DER)

DER for ASN.1, as defined in ITU-T Recommendation X.509, is a more restrictive encoding standard than the
alternative BER (Basic Encoding Rules) for ASN.1, as defined in ITU-T Recommendation X.209, upon which
DER is based. Both BER and DER provide a platform-independent method of encoding objects such as
certificates and messages for transmission between devices and applications.
Domain Name Server (DNS)

Internet service that translates domain names into IP addresses. For instance, the domain name
www.example.com might translate to 198.105.232.4.
Dual-Tone Multi-Frequency (DTMF)

In telephone systems, multi-frequency signalling in which a standard set combinations of two specific voice
band frequencies, one from a group of four low frequencies and the other from a group of four higher
frequencies, are used. Although some military telephones have 16 keys, telephones using DTMF usually have
12 keys. Each key corresponds to a different pair of frequencies. Each pair of frequencies corresponds to one
of the ten decimal digits, or to the symbol “#” or “*”, the “*” being reserved for special purposes.
Dynamic Host Configuration Protocol (DHCP)

TCP/IP protocol that enables PCs and workstations to get temporary or permanent IP addresses (out of a pool)
from centrally-administered servers.
Echo Cancellation
Technique that allows for the isolation and filtering of unwanted signals caused by echoes from the main
transmitted signal.
Far End Disconnect

Refers to methods for detecting that a remote party has hung up. This is also known as Hangup Supervision.
There are several methods that may be used by a PBX/ACD/CO to signal that the remote party has hung up,
including cleardown tone, or a wink.
Federal Communications Commission (FCC)

U.S. government regulatory body for radio, television, interstate telecommunications services, and
international services originating in the United States.
Firewall
A firewall in a networked environment prevents some communications forbidden by the security policy. It has
the basic task of controlling traffic between different zones of trust. Typical zones of trust include the Internet
(a zone with no trust) and an internal network (a zone with high trust).
Foreign Exchange Office (FXO)

A network-provided service in which a telephone in a given local exchange area is connected, via a private
line, to a central office in another, i.e., “foreign”, exchange, rather than the local exchange area’s central office.
This is the office end of an FX circuit (frequently a PBX).
Foreign Exchange Service/Station (FXS)

A network-provided service in which a telephone in a given local exchange area is connected, via a private
line, to a central office in another, i.e., “foreign”, exchange, rather than the local exchange area’s central office.
This is the station (telephone) end of an FX circuit. An FXS port will provide dial tone and ring voltage.
Full-Duplex Connection
Refers to a transmission using two separate channels for transmission and reception and that can transmit in
both ways at the same time. See also Half-Duplex Connection.

G.703
ITU-T recommendation for the physical and electrical characteristics of hierarchical digital interfaces at rates
up to 140Mbit/s.
G.704
ITU-T recommendation for synchronous frame structures on G.703 interfaces up to 45Mbit/s. The
conventional use of G.704 on a 2Mbit/s primary rate circuit provides 30 discrete 64kbit/s channels, with a
further 64kbit/s channel available for common channel signalling.
G.711
Algorithm designed to transmit and receive A-law PCM (Pulse Code Modulation) voice at digital bit rates of 48
kbps, 56 kbps, and 64 kbps. It is used for digital telephone sets on digital PBX and ISDN channels.
G.723.1
A codec that provides the greatest compression, 5.3 kbps or 6.3 kbps; typically specified for multimedia
applications such as H.323 videoconferencing.
G.726
An implementation of ITU-T G.726 standard for conversion linear or A-law or µ-law PCM to and from a 40, 32,
24 or 16 kbit/s channel.
G.729
A codec that provides near toll quality at a low delay which uses compression to 8 kbps (8:1 compression rate).
Gateway
A device linking two different types of networks that use different protocols (for example, between the packet
network and the Public Switched Telephone Network).
Half-Duplex Connection
Refers to a transmission using the same channel for both transmission and reception therefore it can't transmit
and receive at the same time. See also Full-Duplex Connection.
Hunt Group
The hunt group hunts an incoming call to multiple interfaces. It accepts a call routed to it by a routing table or
directly from an interface and creates another call that is offered to one of the configured destination interfaces.
If this destination cannot be reached, the hunt group tries another destination until one of the configured
destinations accepts the call. When an interface accepts a call, the interface hunting is complete and the hunt
group service merges the original call with the new call to the interface that accepted the call.
Impedance
Impedance is the apparent resistance, in an electric circuit, to the flow of an alternating current, analogous to
the actual electrical resistance to a direct current, being the ratio of electromotive force to the current.
Information Transfer Capability (ITC)

A request to the network exchange equipment to ask if a particular type of encoding is allowed. It is also
called ISDN bearer capability or ISDN service.
Integrated Services Digital Network (ISDN)

A set of digital transmission protocols defined by the international standards body for telecommunications, the
ITU-T (formerly called the CCITT). These protocols are accepted as standards by virtually every
telecommunications carrier all over the world.
ISDN complements the traditional telephone system so that a single pair of telephone wires is capable of
carrying voice and data simultaneously. It is a fully digital network where all devices and applications present
themselves in a digital form.

International Telecommunication Union (ITU)

Organization based in Geneva, Switzerland, that is the most important telecom standards-setting body in the
world.
Internet-Drafts
Internet-Drafts are working documents of the IETF, its areas, and its working groups. Note that other groups
may also distribute working documents as Internet-Drafts.
Internet Protocol (IP)

A standard describing software that keeps track of the Internet’s addresses for different nodes, routes outgoing
messages, and recognizes incoming messages.
IP Forwarding
Allows the packet to be forwarded to a specific network based on the packet’s criteria (source IP address and
source Ethernet link).
Jitter
A distortion caused by the variation of a signal from its references which can cause data transmission errors,
particularly at high speeds.
Light Emitting Diode (LED)

A semiconductor diode that emits light when a current is passed through it.
Local Area Network (LAN)

Data-only communications network confined to a limited geographic area, with moderate to high data rates.
See also WAN.
Local Firewall
Allows you to dynamically create and configure rules to filter incoming packets with the unit as destination. The
traffic is analyzed and filtered by all the rules configured.
Management Information Base (MIB)

Specifications containing definitions of management information so that networked systems can be remotely
monitored, configured and controlled.
Media Access Control (MAC) Address

A layer 2 address, 6 bytes long, associated with a particular network device; used to identify devices in a
network; also called hardware or physical address.
Mu (µ)-Law
The PCM (Pulse Code Modulation) voice coding and companding standard used in Japan and North America.
See also A-Law.
Music on Hold (MoH)

Refers to the practice of playing pre-recorded music to fill the silence that would be heard by telephone callers
who have been placed on hold. It is especially common in situations involving customer service.
Network
A group of computers, terminals, and other devices and the hardware and software that enable them to
exchange data and share resources over short or long distances. A network can consist of any combination
of local area networks (LAN) or wide area networks (WAN).

Network Address Translation (NAT)

NAT, also known as network masquerading or IP masquerading, rewrites the source and/or destination
addresses/ports of IP packets as they pass through a router or firewall. It is most commonly used to connect
multiple computers to the Internet (or any other IP network) by using one IP address. This allows home users
and small businesses to cheaply and efficiently connect their network to the Internet. The basic purpose of
NAT is to multiplex traffic from the internal network and present it to the Internet as if it was coming from a
single computer having only one IP address.
There are two types of NAT rules:
 Source rules: They are applied on the source address of outgoing packets.
 Destination rules: They are applied on the destination address of incoming packets.
Network Firewall
Allows dynamically creating and configuring rules to filter packets forwarded by the unit. Since this is a network
firewall, rules only apply to packets forwarded by the unit. The traffic is analyzed and filtered by all the rules
configured.
Off-hook
A line condition caused when a telephone handset is removed from its cradle.
On-hook
A line condition caused when a telephone handset is resting in its cradle.
Packet
Includes three principal elements: control information (such as destination, origin, length of packet), data to be
transmitted, and error detection. The structure of a packet depends on the protocol.
Plain Old Telephone System (POTS)

Standard telephone service used by most residential locations; basic service supplying standard single line
telephones, telephone lines, and access to the public switched network.
Point to Point Protocol over Ethernet (PPPoE)

A proposal specifying how a host personal computer interacts with a broadband modem (i.e., DSL, cable,
wireless, etc.) to access the growing number of Highspeed data networks. Relying on two widely accepted
standards, Ethernet and the point-to-point protocol (PPP), the PPPoE implementation requires virtually no
more knowledge on the part of the end user other than that required for standard Dialup Internet access. In
addition, PPPoE requires no major changes in the operational model for Internet Service Providers (ISPs) and
carriers. The base protocol is defined in RFC 2516.
Port
Network access point, the identifier used to distinguish among multiple simultaneous connections to a host.
Portable Operating System Interface (POSIX)

POSIX is a set of standard operating system interfaces based on the UNIX operating system. The need for
standardization arose because enterprises using computers wanted to be able to develop programs that could
be moved among different manufacturer's computer systems without having to be recoded.
Primary Rate Interface (PRI)

A telecommunications standard for carrying multiple DS0 voice and data transmissions between two physical
locations. All data and voice channels are (ISDN) and operate at 64 kbit/s.
North America and Japan use a T1 of 23 B channels and one D channel which corresponds to a T1 line.
Europe, Australia and most of the rest of the world use the slightly higher capacity E1, which is composed of
31 B channels and one D channel.
Fewer active B channels (also called user channels) can be used for a fractional T1. More channels can be
used with more T1's, or with a fractional or full T3 or E3.

Presentation Indicator (PI)

An information element (IE) field that determines whether a caller’s CLI can be displayed on a Caller ID
device or otherwise presented to the called party.
Private Branch Exchange (PBX)

A small to medium sized telephone system and switch that provides communications between onsite
telephones and exterior communications networks.
Protocol
A formal set of rules developed by international standards bodies, LAN equipment vendors, or groups
governing the format, control, and timing of network communications. A set of conventions dealing with
transmissions between two systems. Typically defines how to implement a group of services in one or two
layers of the OSI reference model. Protocols can describe low-level details of machine-to-machine interfaces
or high-level exchanges between allocation programs.
Proxy Server
An intermediary program that acts as both a server and a client for the purpose of making requests on behalf
of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other
servers. A proxy interprets, and, if necessary, rewrites a request message before forwarding it.
Public Switched Telephone Network (PSTN)

The local telephone company network that carries voice data over analog telephone lines.
QSIG
QSIG is an ISDN based signaling protocol for signaling between private branch exchanges (PBXs) in a Private
Integrated Services Network (PISN). It makes use of the connection-level Q.931 protocol and the application-
level ROSE protocol. ISDN "proper" functions as the physical link layer.
Quality of Service (QoS)

Measure of the telephone service quality provided to a subscriber. This could be, for example, the longest time
someone should wait after picking up the handset before they receive dial tone (three seconds in most U.S.
states).
Real Time Control Protocol (RTCP)

RTCP is the control protocol designed to work in conjunction with RTP. It is standardized in RFC 1889 and
1890. In an RTP session, participants periodically send RTCP packets to convey feedback on quality of data
delivery and information of membership.
Realtime Transport Protocol (RTP)

An IETF standard for streaming realtime multimedia over IP in packets. Supports transport of real-time data
like interactive voice and video over packet switched networks.
Registrar Server
A server that accepts REGISTER requests. A registrar is typically co-located with a proxy or redirect server
and MAY offer location services.
Request for Comment (RFC)

A formal document from the IIETF that is the result of committee drafting and subsequent review by interested
parties. Some RFCs are informational in nature. Of those that are intended to become Internet standards, the
final version of the RFC becomes the standard and no further comments or changes are permitted. Change
can occur, however, through subsequent RFCs that supersede or elaborate on all or parts of previous RFCs.
Screening Indicator (SI)

A service provided by ISDN that can be used to test the trustworthiness of the calling party’s number. This
signalling-related information element is found in octet 3a of the ISDN SETUP message.

Server
A computer or device on a network that works in conjunction with a client to perform some operation.
Session Description Protocol (SDP)

Describes multimedia sessions for the purpose of session announcement, session invitation and other forms
of multimedia session initiation. SDP communicates the existence of a session and conveys sufficient
information to enable participation in the session. SDP is described in RFC 2327.
Session Initiation Protocol (SIP)

A protocol for transporting call setup, routing, authentication, and other feature messages to endpoints within
the IP domain, whether those messages originate from outside the IP cloud over SCN resources or within the
cloud.

A standard of network management that uses a common software agent to manage local and wide area
network equipment from different vendors; part of the Transmission Control Protocol / Internet Protocol (TCP/
IP) suite and defined in RFC 1157.
Simple Network Time Protocol (SNTP)

SNTP, which is an adaptation of the Network Time Protocol (NTP), is widely used to synchronize computer
clocks in the global Internet. It provides comprehensive mechanisms to access national time and frequency
dissemination services, organize the time-synchronization subnet and adjust the local clock in each
participating subnet peer. In most places of the Internet of today, NTP provides accuracies of 1-50 ms,
depending on the characteristics of the synchronization source and network paths.
Subnet
An efficient means of splitting packets into two fields to separate packets for local destinations from packets
for remote destinations in TCP/IP networks.
Switched Circuit Network (SCN)

A communication network, such as the public switched telephone network (PSTN), in which any user may be
connected to any other user through the use of message, circuit, or packet switching and control devices.
T.38
An ITU-T Recommendation for Real-time fax over IP. T.38 addresses IP fax transmissions for IP-enabled fax
devices and fax gateways, defining the translation of T.30 fax signals and Internet Fax Protocols (IFP) packets.
Telephony
The science of translating sound into electrical signals, transmitting them, and then converting them back into
sound.
Transmission Control Protocol/Internet Protocol (TCP/IP)

The basic communication language or protocol of the Internet. It can also be used as a communications
protocol in a private network (either an intranet or an extranet).
Trivial File Transfer Protocol (TFTP)

A simplified version of FTP that transfers files but does not provide password protection, directory capability,
or allow transmission of multiple files with one command.
User Datagram Protocol (UDP)

An efficient but unreliable, connectionless protocol that is layered over IP, as is TCP. Application programs are
needed to supplement the protocol to provide error processing and retransmission of data. UDP is an OSI
layer 4 protocol.

Virtual LAN (VLAN)

A network of computers that behave as if they are connected to the same wire even though they may actually
be physically located on different segments of a LAN. One of the biggest advantages of VLANs is that when
a computer is physically moved to another location, it can stay on the same VLAN without any hardware
reconfiguration.
Virtual Private Network (VPN)

A private communications network usually used within a company, or by several different companies or
organizations, to communicate over a public network. VPN message traffic is carried on public networking
infrastructure (e.g. the Internet) using standard (often insecure) protocols, or over a service provider's network
providing VPN service guarded by well defined Service Level Agreement (SLA) between the VPN customer
and the VPN service provider.
Voice Over IP (VoIP)

The technology used to transmit voice conversations over a data network using the Internet Protocol. Such
data network may be the Internet or a corporate Intranet.
Wide Area Network (WAN)

A large (geographically dispersed) network, usually constructed with serial lines, that covers a large
geographic area. A WAN connects LANs using transmission lines provided by a common carrier.


A P P E N D I X
F List of Acronyms
AC Access Concentrator
AES Advanced Encryption Standard
ANI Automatic Number Identification
BRI Basic Rate Interface
CA Certification Authority
CAS Channel Associated Signalling
CCBS Completion of Calls to Busy Subscriber
CCNR Completion of Calls on No Reply
CHAP Challenge Handshake Authentication Protocol
CLI Command Line Interface
CLIP Calling Line Information Presentation
CLIR Calling Line Information Restriction
CNG Comfort Noise Generator
COLP Connected Line Identification Presentation
COLR Connected Line Identification Restriction
CS-ACELP Conjugate Structure-Algebraic Code Excited Linear Prediction
DER Distinguished Encoding Rules

DNIS Dialed Number Identification Service
DNS Domain Name Server
DSCP Differentiated Services Code Point
DSS1 Digital Subscriber Signaling System No.1
DST Daylight Saving Time
DTMF Dual Tone Multi-Frequency
FQDN Fully Qualified Domain Name

FSK Frequency Shift Keying
GMT Greenwich Mean Time
HTML Hyper Text Markup Language

HTTP HyperText Transfer Protocol
HTTPS HTTP over the Transport Layer Security
Hz Hertz
ICMP Internet Control Message Protocol

IEEE Institute of Electrical & Electronics Engineers
IETF Internet Engineering Task Force
ISDN Integrated Services Digital Network
ITC Information Transfer Capability
ITU International Telecommunication Union
kbps KiloBits Per Second
LAN Local Area Network

LED Light Emitting Diode
MAC Media Access Control

MFC Multi-Frequency Code

Appendix F - List of Acronyms
MIB Management Information Base

MIKEY Multimedia Internet KEYing
MSN Multiple Subscriber Number
MTU Maximum Transmission Unit
NAT Network Address Translation

NBNS NetBIOS Name Server
NPI Numbering Plan Indicator
NT Network Termination
NTP Network Time Protocol
PAP Password Authentication Protocol.

PBX Private Branch eXchange
PEM Privacy Enhanced Mail
PI Presentation Indicator
PPP Point-to-Point Protocol
PPPoE Point-to-Point Protocol over Ethernet
PRACk Provisional Response Acknowledgement
PRI Primary Rate Interface
PSTN Public Switched Telephone Network
QoS Quality of Service
RADIUS Remote Authentication Dial-In User Service

RFC 3711 - The Secure Real-time Transport Protocol (SRTP)
RFC Request For Comment
RTCP Real Time Control Protocol
SCN Switched Circuit Network

SDES Secure Description
SDP Session Description Protocol
SHA Secure Hash Algorithm
SI Screening Indicator
SIP Session Initiation Protocol
SNTP Simple Network Time Protocol
SRTCP Secure Real-Time Transport Control Protocol
SRTP Secure Real-Time Transport Protocol
SSH Secure Socket Shell
SSL Secure Sockets Layer
STD Standard Saving Time
STP Spanning Tree Protocol
TBRL Terminal Balance Return Loss

TCP Transmission Control Protocol
TCP/IP Transmission Control Protocol/Internet Protocol
TE Terminal Equipment
TEI Terminal Endpoint Identifier
TFTP Trivial File Transfer Protocol
TLS Transport Layer Security
TON Type of Number
UDP User Datagram Protocol

UMN Unit Manager Network
UNI User-Network Interface
URI Uniform Resource Identifier
UTC Universal Time Coordinated
VAD Voice Activity Detector

VLAN Virtual Local Area Network
WAN Wide Area Network

WINS Windows Internet Name Service

Appendix F - List of Acronyms

A P P E N D I X
G List of Standards Supported
B
bootp-dhcp-option-88 ................................................................................................................................................... 91, 92
D
draft draft-poetzl-bliss-call-completion-00 ......................................................................................................................... 414
draft-choudhuri-sip-info-digit-00 ................................................................................................................................ 354, 381
draft-ietf-http-authentication-03 ........................................................................................................................ 519, 535, 543
draft-ietf-sipping-realtimefax-00 ........................................................................................................................................ 309
E
ETS 300 207 - Call Diversion and Call Rerouting ............................................................................................................ 209
ETSI 300 659-1 January 2001 (Annex B) ......................................................................................................................... 160
ETSI EN 300659-3 ........................................................................................................................................................... 164
I
IEEE 802.1Q -Virtual Bridged Local Area Networks ........................................................................................................... 98
IEEE 802.1X - Port Based Network Access Control ......................................................................................................... 106
IEEE Std 1003.1-2001 - IEEE Standard for Information Technology---Portable Operating System Interface (POSIX®) 457
ITU-T Recommendation E.164 - The international public telecommunication numbering plan ........................................ 455
ITU-T Recommendation F.69 - List of Telex Destination Codes ...................................................................................... 455
ITU-T Recommendation G.703 - Physical/Electrical Characteristics of Hierarchical Digital Interfaces ............ 182, 193, 219
ITU-T Recommendation G.704 - Synchronous Frame Structures Used at Primary and Secondary Hierarchy Levels ... 182,
193, ............................................................................................................................................................. 219
ITU-T Recommendation G.711 ........................................................................................................................................ 349
ITU-T Recommendation G.723.1 ..................................................................................................................................... 350
ITU-T Recommendation I.430 - Basic user-network interface - Layer 1 specification ...................................................... 193
ITU-T Recommendation I.431 - Primary rate user-network interface - Layer 1 specification ........................................... 182
ITU-T Recommendation Q.24 - Multifrequency push-button signal reception .................................................................. 354
ITU-T Recommendation Q.310-Q.332 - Specifications of Signalling System R1 ............................................................. 251
ITU-T Recommendation Q.421 - Digital line signalling code ............................................................................................ 219
ITU-T Recommendation Q.441 - Signalling code ............................................................................................................. 219
ITU-T Recommendation Q.921 - ISDN user-network interface - Data link layer specification ................................. 182, 193
ITU-T Recommendation Q.931 - ISDN user-network interface layer 3 specification for basic call control ....... 182, 193, 458
ITU-T Recommendation T.38 ................................................................................................................................... 309, 353
ITU-T Recommendation X.121 - International numbering plan for public data networks ................................................. 455
R
RFC 1034 - Domain Names - Concepts and Facilities ....................................................................................................... 90
RFC 1035 - Domain Names - Implementation and Specification ....................................................................................... 90
RFC 1157 - Simple Network Management Protocol (SNMP) ........................................................................................... 559
RFC 1332 - The PPP Internet Protocol Control Protocol (IPCP) ...................................................................................... 103
RFC 1334 - PPP Authentication Protocols ....................................................................................................................... 103
RFC 1350 - The TFTP Protocol (Revision 2) ........................................................................................... 441, 519, 535, 543
RFC 1661 - The Point-to-Point Protocol (PPP) ................................................................................................................ 103
RFC 1877 - PPP Internet Protocol Control Protocol Extensions for Name Server Addresses ......................................... 103
RFC 1886 - DNS Extensions to support IP version 6 ......................................................................................................... 90
RFC 1890 - RTP Profile for Audio and Video Conferences with Minimal Control ............................................................ 354
RFC 1910 - User-based Security Model for SNMPv2 ...................................................................................................... 559
RFC 1945 - Hypertext Transfer Protocol - HTTP/1.0 ......................................................................................................... 41
RFC 1994 - Challenge Handshake Authentication Protocol (CHAP) ............................................................................... 103
RFC 2030 - Simple Network Time Protocol (SNTP) Version 4 for IPv4, IPv6 and OSI ...................................................... 91
RFC 2104 - HMAC Keyed-Hashing for Message Authentication ..................................................................................... 559

Appendix G - List of Standards Supported
RFC 2131 - Dynamic Host Configuration Protocol ................................................................................................... 107, 147

RFC 2132 - DHCP Options and BOOTP Vendor Extensions ................................................................................... 107, 147
RFC 2181 - Clarifications to the DNS Specification ........................................................................................................... 90
RFC 2246 - The TLS Protocol Version 1.0 ............................................................................................... 303, 520, 536, 545
RFC 2459 - X.509 Digital Certificates ............................................................................................................... 520, 536, 545
RFC 2460 - IPv6 ................................................................................................................................................................. 98
RFC 2464 - Transmission of IPv6 Packets over Ethernet Networks .................................................................................. 98
RFC 2475 - An Architecture for Differentiated Services ................................................................................................... 113
RFC 2516 - A Method for Transmitting PPP Over Ethernet (PPPoE) .............................................................................. 103
RFC 2543 - SIP, Session Initiation Protocol ............................................................................................. 279, 285, 287, 319
RFC 2576 - Coexistence between Version 1, Version 2, and Version 3 of the Internet-standard Network Management
Framework .................................................................................................................................................. 559
RFC 2616 - Hypertext Transfer Protocol - HTTP/1.1 ......................................................................... 41, 441, 519, 535, 543
RFC 2617 - HTTP Authentication - Basic and Digest Access Authentication .......................................... 519, 535, 543, 583
RFC 2705 - Media Gateway Control Protocol (MGCP) Version 1.0 ................................................................................. 395
RFC 2741 - Agent Extensibility (AgentX) Protocol Version 1 ........................................................................................... 559
RFC 2818 - HTTP Over TLS ............................................................................................................................ 520, 536, 545
RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals ................................................... 354
RFC 2865 - Remote Authentication Dial In User Service (RADIUS) ................................................................................ 583
RFC 2866 - RADIUS Accounting ...................................................................................................................................... 583
RFC 2976 - The SIP INFO Method .................................................................................................................................. 411
RFC 3164 - The BSD Syslog Protocol ............................................................................................................................... 71
RFC 3261 - SIP, Session Initiation Protocol ..................................................................... 279, 285, 287, 299, 303, 310, 318
RFC 3262 - Reliability of Provisional Responses in the Session Initiation Protocol (SIP) ................................................ 340
RFC 3263 - Session Initiation Protocol (SIP) - Locating SIP Servers .............................................................................. 324
RFC 3264 - An Offer/Answer Model with the Session Description Protocol (SDP) .................. 309, 312, 318, 319, 321, 322
RFC 3265 - Session Initiation Protocol (SIP)-Specific Event Notification ......................................................................... 432
RFC 3268 - Advanced Encryption Standard (AES) Ciphersuites for Transport Layer Security (TLS) ............. 520, 536, 545
RFC 3280 - Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL) Profile ..... 520, 536,
545, ............................................................................................................................................................. 553
RFC 3311 - The Session Initiation Protocol (SIP) UPDATE Method ................................................................................ 340
RFC 3315 - DHCPv6 ........................................................................................................................................................ 107
RFC 3323 - A Privacy Mechanism for the Session Initiation Protocol (SIP) ..................................................................... 488
RFC 3325 - Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks .. 488
RFC 3398 - Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping 332
RFC 3411 - An Architecture for Describing Simple Network Management Protocol (SNMP) Management Frameworks 559
RFC 3412 - Message Processing and Dispatching for the Simple Network Management Protocol (SNMP) .................. 559
RFC 3413 - Simple Network Management Protocol (SNMP) Applications ...................................................................... 559
RFC 3414 - User-based Security Model (USM) for version 3 of the Simple Network Management Protocol (SNMPv3) 559
RFC 3415 - View-based Access Control Model (VACM) for the Simple Network Management Protocol (SNMP) .......... 559
RFC 3416 - Version 2 of the Protocol Operations for the Simple Network Management Protocol (SNMP) ..................... 559
RFC 3417 - Transport Mappings for the Simple Network Management Protocol (SNMP) ............................................... 559
RFC 3442 - The Classless Static Route Option for Dynamic Host Configuration Protocol (DHCP) version 4 ................. 129
RFC 3515 - The Session Initiation Protocol (SIP) Refer Method ..................................................................................... 309
RFC 3550 - RTP - A Transport Protocol for Real-Time Applications ............................................................................... 349
RFC 3551 - RTP Profile for Audio and Video Conferences with Minimal Control ............................................................ 349
RFC 3617 - Uniform Resource Identifier (URI) Scheme for the Trivial File Transfer Protocol ......................... 519, 535, 543
RFC 3711 - The Secure Real-time Transport Protocol (SRTP) ....................................................................................... 373
RFC 3826 - The Advanced Encryption Standard (AES) Cipher Algorithm in the SNMP User-based Security Model ..... 559
RFC 3830 - MIKEY - Multimedia Internet KEYing ............................................................................................................ 373
RFC 3842 - The Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)
..................................................................................................................................................................... 432
RFC 3863 - Presence Information Data Format (PIDF) ................................................................................................... 287
RFC 3903 - Session Initiation Protocol (SIP) Extension for Event State Publication ............................................... 279, 287
RFC 4040 - RTP Payload Format for a 64 kbit/s Transparent Call .......................................................................... 352, 353
RFC 4193 - Unique_local_address .................................................................................................................................... 98
RFC 4235 - An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol (SIP) .................................... 414
RFC 4251 - The Secure Shell (SSH) Protocol Architecture ............................................................................................... 22
RFC 4291 - IP Version 6 Addressing Architecture ............................................................................................................. 98
RFC 4443 - ICMPv6 ........................................................................................................................................................... 98
RFC 4567 - Key Management Extensions for Session Description Protocol (SDP) and Real Time Streaming Protocol
(RTSP) ........................................................................................................................................................ 373
RFC 4568 - SDES - Security Descriptions for Media Streams ......................................................................................... 373
RFC 4579 - Session Initiation Protocol (SIP) - Call Control - Conferencing for User Agents ........................................... 425
RFC 4861 - IPv6_neighbor_discovery ................................................................................................................................ 98
RFC 4862 - IPv6_stateless_autoconf ................................................................................................................................. 98

RFC 5806 - Diversion Indication in SIP ............................................................................................................................ 462

RFC 854 - Telnet Protocol Specification ............................................................................................................................ 21
RFC 959 - File Transfer Protocol ...................................................................................................................... 519, 535, 543
T
TR-069 - CPE WAN Management Protocol v1.1 (Issue 1 Amendment 2) ....................................................................... 565
TR-098 - Internet Gateway Device Data Model for TR-069 (Issue 1 Amendment 2) ....................................................... 565
TR-104 - Provisioning Parameters for VoIP CPE ............................................................................................................. 565
TR-106 - Data Model Template for TR-069-Enabled Devices .......................................................................................... 565
TR-111 - Applying TR-069 to Remote Management of Home Networking Devices ......................................................... 565
B
bootp-dhcp-option-8893, 94
D
draft draft-poetzl-bliss-call-completion-00420
draft-choudhuri-sip-info-digit-00358, 387
draft-ietf-http-authentication-03525, 541, 549
draft-ietf-sipping-realtimefax-00311
E
ETS 300 207 - Call Diversion and Call Rerouting211
ETSI 300 659-1 January 2001 (Annex B)162
ETSI EN 300659-3166
I
IEEE 802.1Q -Virtual Bridged Local Area Networks100
IEEE 802.1X - Port Based Network Access Control108
IEEE Std 1003.1-2001 - IEEE Standard for Information Technology---Portable Operating
System Interface (POSIX®)463
ITU-T Recommendation E.164 - The international public telecommunication numbering
plan461
ITU-T Recommendation F.69 - List of Telex Destination Codes461
ITU-T Recommendation G.703 - Physical/Electrical Characteristics of Hierarchical Digi-
tal Interfaces184, 195, 221
ITU-T Recommendation G.704 - Synchronous Frame Structures Used at Primary and Sec-
ondary Hierarchy Levels184, 195, 221
ITU-T Recommendation G.711353
ITU-T Recommendation G.723.1354
ITU-T Recommendation I.430 - Basic user-network interface - Layer 1 specification195
ITU-T Recommendation I.431 - Primary rate user-network interface - Layer 1 specifica-
tion184
ITU-T Recommendation Q.24 - Multifrequency push-button signal reception358
ITU-T Recommendation Q.310-Q.332 - Specifications of Signalling System R1253
ITU-T Recommendation Q.421 - Digital line signalling code221
ITU-T Recommendation Q.441 - Signalling code221
ITU-T Recommendation Q.921 - ISDN user-network interface - Data link layer specifica-

tion184, 195
ITU-T Recommendation Q.931 - ISDN user-network interface layer 3 specification for ba-
sic call control184, 195, 464
ITU-T Recommendation T.38311, 357
ITU-T Recommendation X.121 - International numbering plan for public data networks461
R
RFC 1034 - Domain Names - Concepts and Facilities92
RFC 1035 - Domain Names - Implementation and Specification92
RFC 1157 - Simple Network Management Protocol (SNMP)563
RFC 1332 - The PPP Internet Protocol Control Protocol (IPCP)105
RFC 1334 - PPP Authentication Protocols105
RFC 1350 - The TFTP Protocol (Revision 2)447, 525, 541, 549
RFC 1661 - The Point-to-Point Protocol (PPP)105
RFC 1877 - PPP Internet Protocol Control Protocol Extensions for Name Server Addresses
105
RFC 1886 - DNS Extensions to support IP version 692
RFC 1890 - RTP Profile for Audio and Video Conferences with Minimal Control358
RFC 1910 - User-based Security Model for SNMPv2563
RFC 1945 - Hypertext Transfer Protocol - HTTP/1.041
RFC 1994 - Challenge Handshake Authentication Protocol (CHAP)105
RFC 2030 - Simple Network Time Protocol (SNTP) Version 4 for IPv4, IPv6 and OSI93
RFC 2104 - HMAC Keyed-Hashing for Message Authentication563
RFC 2131 - Dynamic Host Configuration Protocol109, 149
RFC 2132 - DHCP Options and BOOTP Vendor Extensions109, 149
RFC 2181 - Clarifications to the DNS Specification92
RFC 2246 - The TLS Protocol Version 1.0305, 526, 542, 550
RFC 2459 - X.509 Digital Certificates526, 542, 550
RFC 2460 - IPv6100
RFC 2464 - Transmission of IPv6 Packets over Ethernet Networks100
RFC 2475 - An Architecture for Differentiated Services115
RFC 2516 - A Method for Transmitting PPP Over Ethernet (PPPoE)105
RFC 2543 - SIP, Session Initiation Protocol281, 287, 289, 321
RFC 2576 - Coexistence between Version 1, Version 2, and Version 3 of the Internet-stan-
dard Network Management Framework563
RFC 2616 - Hypertext Transfer Protocol - HTTP/1.141, 447, 525, 541, 549
RFC 2617 - HTTP Authentication - Basic and Digest Access Authentication525, 541, 549, 591
RFC 2705 - Media Gateway Control Protocol (MGCP) Version 1.0401
RFC 2741 - Agent Extensibility (AgentX) Protocol Version 1563
RFC 2818 - HTTP Over TLS526, 542, 550
RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals358
RFC 2865 - Remote Authentication Dial In User Service (RADIUS)591
RFC 2866 - RADIUS Accounting591
RFC 2976 - The SIP INFO Method417
RFC 3164 - The BSD Syslog Protocol71
RFC 3261 - SIP, Session Initiation Protocol281, 287, 289, 301, 305, 312, 320
RFC 3262 - Reliability of Provisional Responses in the Session Initiation Protocol (SIP)342

RFC 3263 - Session Initiation Protocol (SIP) - Locating SIP Servers326

RFC 3264 - An Offer/Answer Model with the Session Description Protocol (SDP)311, 314,
321, 323, 324
RFC 3265 - Session Initiation Protocol (SIP)-Specific Event Notification437
RFC 3268 - Advanced Encryption Standard (AES) Ciphersuites for Transport Layer Secu-
rity (TLS)526, 542, 550
RFC 3280 - Internet X.509 Public Key Infrastructure Certificate and Certificate Revoca-
tion List (CRL) Profile526, 542, 550, 557
RFC 3311 - The Session Initiation Protocol (SIP) UPDATE Method342
RFC 3315 - DHCPv6109
RFC 3323 - A Privacy Mechanism for the Session Initiation Protocol (SIP)494
RFC 3325 - Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity
within Trusted Networks494
RFC 3398 - Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Ini-
tiation Protocol (SIP) Mapping334
RFC 3411 - An Architecture for Describing Simple Network Management Protocol (SN-
MP) Management Frameworks563
RFC 3412 - Message Processing and Dispatching for the Simple Network Management
Protocol (SNMP)563
RFC 3413 - Simple Network Management Protocol (SNMP) Applications563
RFC 3414 - User-based Security Model (USM) for version 3 of the Simple Network Man-
agement Protocol (SNMPv3)563
RFC 3415 - View-based Access Control Model (VACM) for the Simple Network Manage-
ment Protocol (SNMP)563
RFC 3416 - Version 2 of the Protocol Operations for the Simple Network Management
Protocol (SNMP)563
RFC 3417 - Transport Mappings for the Simple Network Management Protocol (SNMP)
563
RFC 3442 - The Classless Static Route Option for Dynamic Host Configuration Protocol
(DHCP) version 4131
RFC 3515 - The Session Initiation Protocol (SIP) Refer Method311
RFC 3550 - RTP - A Transport Protocol for Real-Time Applications353
RFC 3551 - RTP Profile for Audio and Video Conferences with Minimal Control353
RFC 3617 - Uniform Resource Identifier (URI) Scheme for the Trivial File Transfer Pro-
tocol525, 541, 549
RFC 3711 - The Secure Real-time Transport Protocol (SRTP)377
RFC 3826 - The Advanced Encryption Standard (AES) Cipher Algorithm in the SNMP
User-based Security Model563
RFC 3830 - MIKEY - Multimedia Internet KEYing377
RFC 3842 - The Message Summary and Message Waiting Indication Event Package for
the Session Initiation Protocol (SIP)437
RFC 3863 - Presence Information Data Format (PIDF)289
RFC 3903 - Session Initiation Protocol (SIP) Extension for Event State Publication281, 289
RFC 4040 - RTP Payload Format for a 64 kbit/s Transparent Call356, 357
RFC 4193 - Unique_local_address100
RFC 4235 - An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol
(SIP)420

RFC 4251 - The Secure Shell (SSH) Protocol Architecture22

RFC 4291 - IP Version 6 Addressing Architecture100
RFC 4443 - ICMPv6100
RFC 4567 - Key Management Extensions for Session Description Protocol (SDP) and Real
Time Streaming Protocol (RTSP)377
RFC 4568 - SDES - Security Descriptions for Media Streams377
RFC 4579 - Session Initiation Protocol (SIP) - Call Control - Conferencing for User Agents
430
RFC 4861 - IPv6_neighbor_discovery100
RFC 4862 - IPv6_stateless_autoconf100
RFC 5806 - Diversion Indication in SIP468
RFC 854 - Telnet Protocol Specification21
RFC 959 - File Transfer Protocol525, 541, 549
T
TR-069 - CPE WAN Management Protocol v1.1 (Issue 1 Amendment 2)569
TR-098 - Internet Gateway Device Data Model for TR-069 (Issue 1 Amendment 2)569
TR-104 - Provisioning Parameters for VoIP CPE569
TR-106 - Data Model Template for TR-069-Enabled Devices569
TR-111 - Applying TR-069 to Remote Management of Home Networking Devices569

Index
Numerics bearer capabilities mapping, vs. codec 359
preferred codec choice 317
10 BaseT BRI interface configuration
defined 711 bypass connection 205
100 BaseT calling name max length 200
defined 711 channel allocation strategy 199
802.1q, in local QoS 117 clock mode 197
clock reference 59
A connection type 197
copy config to all interfaces 196
AAA, service encoding scheme, fallback 198
user access 17, 18
encoding scheme, preferred 198
access control
endpoint type 196
RADIUS 592 exclusive B-channel selection 200
accounting type 593
inband DTMF dialing 199
authentication type 592
inband tone generation 199
servers configuration 593 maximum active calls 199
type 592
network location 198
user
overlap dialing 200
access rights 591 restart on startup 200
add 591
select interface 196
delete 591
sending complete information 200
modify 591 signal information element 199
access rights 591
signalling protocol 198
description 595
bridging, hardware configuration 63
RADIUS servers 594
bypass
access, CLI keyword 27 activation 167
accounting type, in RADIUS 593
activation DTMF map 168
ACK branch matching, in SIP 328
deactivation timeout 168
acronyms 721 Bypass connection 6
activate licence 599
bypass connection, in BRI 205
A-Law 353
defined 711
alias, CLI keyword 28 C
answering on caller ID, on FXO port 170 call
AOC-D support, ISDN interface 214, 215 automatic 419
AOC-E support, ISDN interface 214, 215 completion
audience, intended xxv CCBS 420
authentication information 301 CCBS, using 423
creating 302 CCNR 420
deleting 304 CCNR, using 423
editing 302 emergency
moving 304 enabling 407
authentication type, in RADIUS 592 emergency override 439, 441
auto cancel timeout, ISDN interface 210 forward
auto-configuration on busy 409, 411
E&M interfaces 255 on no answer 412, 413
ISDN interfaces 180 unconditional 414, 415
R2 interfaces 223 hold 435
automatic direction attributes 321
call 419 putting on hold 427, 436
configuration update second 436, 437
number of retries 539 transfer
on restart 536 attended 424, 425
specific time interval 537 blind 424, 425
auto-routing waiting 426
auto-routable parameter 519 disabling 427
enabling 517 enabling 427
introduction 517 IMS-3GPP Communication Waiting 427
SIP gateway 519 using 427
Call Detail Record 456
B call failure timeout, on FXO port 172
call properties translation, in call routing
base ports 397

Index
built from 503 Called SIP username 468

creating 502 Called TON 465
defined 502 Called URI 466
deleting 504 Calling Bearer Channel 468
editing 502 Calling E164 465
modified property 503 Calling Host 466
moving 504 Calling ITC 467
name 502 Calling Name 466
call rejection (drop) causes 508 Calling NPI 465
call router Calling Phone Context 468
call properties translation Calling PI 466
built from 503 Calling SI 466
creating 502 Calling SIP username 468
defined 502 Calling TON 465
deleting 504 Calling Uri 466
editing 502 Date/Time 467
modified property 503 Last Diverting E.164 469
moving 504 Last Diverting Number Presentation 470
name 502 Last Diverting Party Number Type 469
hairpinning 515 Last Diverting Private Type of Number 469
hunts Last Diverting Public Type of Number 469
call rejection (drop) causes 508 Last Diverting Reason 468
creating 505 Original Diverting E.164 469
defined 505 Original Diverting Number Presentation 470
deleting 512 Original Diverting Party Number Type 469
destinations 506 Original Diverting Private Type of Number 469
editing 505 Original Diverting Public Type of Number 469
moving 512 Original Diverting Reason 468
selection algorithm 506 special tags 470
timeout 506 signalling properties
introduction 461 180 with SDP 496
limitations 462 183 without SDP 496
mappings call properties translation 497
defined 484 creating 495
mappings, expression defined 494
accessing 486 deleting 498
criteria 487 destination host 496
deleting 494 early connect 495
moving 494 early disconnect 496
sub mappings 493 editing 495
transformation 490 moving 498
mappings, type name 495
accessing 485 privacy level 496
input call property 486 SIP headers translation 497
moving 494 SIP headers translation
transformation 486 built from 500
regular expression 463 creating 499
routes defined 498
call property to compare 481 deleting 501
creating 480 editing 499
defined 479 modified header 499
deleting 484 moving 501
destination, of call 483 name 499
editing 480 SIP redirects
expression to compare 482 creating 513
mapping, apply 483 defined 513
moving 484 deleting 514
signalling, of call 483 editing 513
source to compare 481 moving 514
routing type status information 478
Called Bearer Channel 468 call waiting
Called E164 465 private number for SIP INFO 320
Called Host 466 Called Bearer Channel, routing type 468
Called Name 466 Called E164, routing type 465
Called NPI 465 Called Host, routing type 466
Called Phone Context 468 Called Name, routing type 466

Called NPI, routing type 465 limitation 372

Called Phone Context, routing type 468 Clear Mode
Called SIP username, routing type 468 defined 356
Called TON, routing type 465 enabling
Called URI, routing type 466 for data transmission 367
caller ID for voice transmission 367
customization, in POTS 160 packetization time 367
DTMF signalling 162 payload type 367
FSK generation 162 priority
generation 162 for data codecs 367
Calling Bearer Chanel, routing type 468 for voice codecs 367
Calling E164, routing type 465 CLI
Calling Host, routing type 466 accessing via Console port 20
Calling ITC, routing type 467 accessing via SSH 22
calling name max length, ISDN interface accessing via Telnet 21
BRI configuration 200 command completion 24
PRI configuration 190 contexts 23
Calling Name, routing type 466 current unit status 25
Calling NPI, routing type 465 defined 19
Calling Phone Context, routing type 468 help 25
Calling PI, routing type 466 history 24
Calling SI, routing type 466 keywords
Calling SIP username, routing type 468 access 27
Calling TON, routing type 465 alias 28
Calling Uri, routing type 466 cd 29
CAS. see R2 configuration columnars 29
CCBS commands 29
setting up 420 defval 30
using 423 dump 30
CCNR get 31
setting up 420 help 32
using 423 indexes 33
cd, CLI keyword 29 keys 33
CDR (Call Detail Record) 456 logs off 34
CED tone detection 396 logs on 34
behaviour 396 ls 34
certificate name 35
validation, in TLS 318 objects 35
certificates pcapture 36
deleting 557 ping 36, 37
transferring 560 scalars 37
upload through web browser 558 set 37
validation, in firmware pack update 554 show 38
channel allocation strategy sysinfo 39
E&M interface, configuration 258 tables 39
ISDN interface type 39
BRI configuration 199 unalias 28
PRI configuration 189 macros 24
R2 interface, configuration 226 service restart 24
channel range syslog messages 25
E&M interface, configuration 258 welcome message 25
ISDN interface CLIP, ISDN interface 212
PRI configuration 188 CLIR, ISDN interface 212
R2 interface, configuration 226 override 213
Clear Channel clock mode
defined 356 E&M interface, configuration 257
enabling ISDN interface
for data transmission 368 BRI configuration 197
for voice transmission 368 PRI configuration 186
packetization time 368 R2 interface, configuration 225
payload type 368 clock reference
priority PRI configuration 61
for data codecs 368 clock reference, in ISDN
for voice codecs 368 BRI configuration 59
clear channel fax CNG tone detection 396, 455
call flow 371 codec

Index
bearer capabilities mapping 359 for data codecs 370

preferred codec choice 317 for voice codecs 370
Clear Channel COLP, ISDN interface 213
defined 356 COLR, ISDN interface 213
enabling override 214
for data transmission 368 columnars, CLI keyword 29
for voice transmission 368 commands, CLI keyword 29
packetization time 368 conference call
payload type 368 limitation 429
priority configuration
for data codecs 368 executing script
for voice codecs 368 from a PC 535
Clear Mode from server 531
defined 356 exporting to script 528
enabling to a PC 529
for data transmission 367 to a URL 528
for voice transmission 367 configuration backup image
packetization time 367 backing up 541
payload type 367 decryption 541
priority location path 541
for data codecs 367 privacy key 541
for voice codecs 367 remote server
G.711 HTTP server, configuring 542
defined 353 HTTPS server, configuring 542
enabling SNTP server, configuring 541
for data transmission 361 TFTP server, configuring 541
for voice transmission 361 restoring 541
packetization time 361 server, IP address 541
priority transfer protocols 541
for data transmission 361 configuration script
for voice transmission 361 executing configuration
G.723 from a PC 535
bit rate 362 from server 531
defined 354 exporting configuration 528
enabling for voice transmission 362 to a PC 529
packetization time 363 to a URL 528
priority for voice codecs 362 configuration script download
G.726 automatic update
defined 354 number of retries 539
enabling on restart 536
for data transmission 363 specific time interval 537
for voice transmission 363 decryption 531
packetization time 365 DHCP auto-provisioning
payload type 365 DHCP
priority auto-provisioning (scripts) 538
for data transmission 363 downloading 536
for voice transmission 363 location path 531
G.729 privacy key 531
defined 355 remote server
enabling for voice transmission 365 HTTP server, configuring 526
packetization time 366 HTTPS server, configuring 526
priority for voice codecs 365 SNTP server, configuring 526
voice activity detection 366 TFTP server, configuring 525
T.38 script example 540
defined 357 server, IP address 531
enabling 373 transfer protocols 531
number of redundancy packets 373 connection type, ISDN interface
priority 373 BRI configuration 197
X-CCD Clear Channel Console port
defined 357 accessing CLI via 20
enabling contact domain, in SIP 290
for data transmission 370 Content-type, unsupported 332
for voice transmission 370 Country 166
packetization time 370 country selection, R2 interface 227
payload type 370 country-specific parameters
priority CNG tone detection 455

country 451 supported queries 348

custom tone configuration 443 DNS SRV
dialing settings 453 record lock, in SIP 326
list of countries 603 document
user gain conventions xxix
input 452 intended audience xxv
output 452 objectives xxiv
custom tones, configuring 443 structure xxv
CWMP domain name
ACS parameters 572 server, in embedded DHCP server 152
Allow Unauthenticated UDP Connection Requests 579 download, firmware. see firmware pack update
general configuration 571 DTMF
parameter type validation 580, 581 base ports 397
periodic inform parameters 573 defined 713
profile creation 577 detection 392
TR-069 configuration 575 duration value 454
TR-104 configuration 576 machine detection
TR-106 configuration 577 CED tone detection 396
TR-111 577 CED tone detection, behaviour 396
CNG tone detection 396
V.21 modulation detection 396
D out-of-band 391
Date/Time, routing type 467 signalling, caller ID 162
decryption transport type
configuration backup image 541 over the SIP protocol 392
configuration script download 531 DTMF map
default gateway allowed 405
client, setting 91 defined
server, in embedded DHCP server 152 # and * characters 403
default router. see default gateway combining two expressions 402
defval, CLI keyword 30 special characters 402
delayed hot line 433 timer 403
detection custom repetition, on FXO port 174 using 402
DHCP connection type 100 validating 403
DHCP server definition 401
configuring 109 general parameters 404
embedded refused 407
default gateway 152 dump, CLI keyword 30
defined 149
DNS servers 153
domain name server 152 E
lease time 151 E&M
leases 155 partial reset 256
NBNS servers 155 E&M configuration
NTP servers 154 auto-configuration 255
subnets, managing 151 channel allocation strategy 258
DHCPv4 classless static route option 131 channel range 258
diagnostic traces, enabling 71 clock mode 257
PCM capture 74 digit timers 265
dial tone detection mode, on FXO port 169 encoding scheme 259
dial tone detection timeout, on FXO port 169 encoding scheme, preferred 259
dialing introduction 253
settings IP synchronization 270
delay between MFR1s 454 enabling 270
DTMF duration value 454 master units 270
inter-digit dial delay 454 resetting synchronization data 271
Differentiated Services (DS) Field, in local QoS 115 slave units 271
direct IP address line coding 258
call 434 line framing 258
direction attribute "sendonly" 317, 323 link timers 267
direction attributes, in SIP 321 maximum active calls 259
disabling endpoints 67 preset 255
distinctive ring 605 PRI statistics 272
DNS selecting signalling protocol 254
client 92 signalling 260
server, in embedded DHCP server 153 B bit setting 261

Index
C bit setting 261 PRI configuration 190

D bit setting 261
override default settings 261
timers 263
F
tones 268 facility ISDN services
emergency call enabling 211
enabling 407 factory reset
enabling endpoints 67 reverting to 16
encoding scheme see also partial reset
E&M interface, configuration 259 fax
R2 interface, configuration 227 call waiting tone, disabling 427
encoding scheme, ISDN interface calling tone detection, enabling 455
fallback clear channel
BRI configuration 198 call flow 371
PRI configuration 188 limitation 372
preferred T.38
BRI configuration 198 defined 357
PRI configuration 188 enabling 373
encoding scheme, preferred number of redundancy packets 373
E&M interface, configuration 259 priority 373
R2 interface, configuration 227 user gain vs communication quality 452
endpoint file manager 597
administration 68 firewall, local
locking/unlocking 67 default policy 121
unit shutting down behsviour 69 disabling 125
unregistered, behaviour when 296 partial reset 121
endpoint services rule, creating 122
automatic call 419 rule, deleting 125
call completion rule, editing 122
CCBS 420 rule, moving 124
CCNR 420 status information 124
call forward firewall, network
on busy 409 default policy 135
on no answer 412 disabling 139
unconditional 414 rule, creating 136
call transfer rule, deleting 139
attended transfer 424 rule, editing 136
blind transfer 424 rule, moving 139
call waiting 426 status information 139
conference call 429 firmware pack update
limitation 429 certificate validation 554
direct IP address call 434 downgrade 556
emergency call override 439, 441 firmware pack, defined 549
hold 435 launching 555
direction attributes 321 remote server
hook-flash processing 418 HTTP server, configuring 550
second call 436 HTTPS server, configuring 550
endpoint type, ISDN interface SNTP server, configuring 550
BRI configuration 196 TFTP server, configuring 550
PRI configuration 185 rollback 552
error causes mapping, setting 334 Spanning Tree Protocol 556
cause to SIP 338 transfer protocols 553
SIP to cause 336 update files path 553
Ethernet connection update files server
setting speed of 111 IP address 553
Ethernet, link configuration 108 port number 553
event handling, in SIP 348 zip file, extracting 555
events flash hook detection range 166
monitoring parameters force end of call on call failure, on FXO port 171
monitoring parameters, in events 79 forced end of call on silence detection mode, on FXO port 172
notification forced end of call on tone detection mode, on FXO port 172
notification events 77 Foreign Exchange Office (FXO)
rule defined 713
deleting 79 see also endpoint
exclusive B-channel selection, ISDN interface Foreign Exchange Service/Station (FXS)
BRI configuration 200 defined 713

see also endpoint defined 355

forked provisional reponses behaviour, in SIP 343 enabling for voice transmission 365
FSK generation, caller ID 162 packetization time 366
FXO configuration priority for voice codecs 365
answering on caller ID 170 voice activity detection 366
call failure timeout 172 gateways, multiple SIP
detection custom repetition 174 adding 277
dial tone detection mode 169 keep alive destinations 286
dial tone detection timeout 169 messaging servers 284
force end of call on call failure 171 proxy servers 284
forced end of call on silence detection mode 172 registrar servers 283
forced end of call on tone detection mode 172 removing 277
link state verification 171 get, CLI keyword 31
link state verification timeout 171 group port management 67
not allowed behavior 170
pre dial delay 169
silence detection timeout 172
H
tone detection custom cadence 173 hairpinning 515
tone detection custom frequency 173 hardware configuration
wait before answering delay 170 BRI interface
wait for callee to answer 170 clock reference 59
FXS power feeding over ISDN 62
auto cancel timeout 163 bridging 63
bypass PRI interface
activation 167 clock reference 61
activation DTMF map 168 line type 61, 222, 254
deactivation timeout 168 signaling 61, 222, 254
country override header, SIP user agent
flash hook detection range 166 sending 299
loop current 166 help, CLI keyword 32
disconnect delay 163 hold, putting a call on 427, 435, 436
inband ringback 164 direction attributes 321
line supervision mode 163 home domain proxy override 345
override country customization 165 hook-flash processing 418
power drop on disconnect duration 164 HTTP
service activation 164 remote server, configuring 447, 526, 542, 550
shutdown behavior 164 HTTP User-Agent Header
format 42
HTTPS
G remote server, configuring 526, 542, 550
G.711 hunts, in call routing
defined 353 call rejection (drop) causes 508
enabling creating 505
for data transmission 361 defined 505
for voice transmission 361 deleting 512
packetization time 361 destinations 506
priority editing 505
for data transmission 361 moving 512
for voice transmission 361 selection algorithm 506
G.723 timeout 506
bit rate 362
defined 354
enabling for voice transmission 362 I
packetization time 363 IEEE 802.1q, in local QoS 117
priority for voice codecs 362 IMS-3GPP Communication Waiting 427
G.726 inband DTMF dialing, ISDN interface
defined 354 BRI configuration 199
enabling PRI configuration 189
for data transmission 363 inband tone generation, ISDN interface
for voice transmission 363 BRI configuration 199
packetization time 365 PRI configuration 189
payload type 365 indexes, CLI keyword 33
priority InformationFollowing operation, ISDN interface
for data transmission 363 PRI configuration 194
for voice transmission 363 intended audience xxv
G.729 inter-digit dial delay 454

Index
IP address signalling protocol 198

configuration backup server 541 interop parameters
configuration script download server 531 copy config to all interfaces 206, 210, 211
firmware pack update files server 553 maximum facility waiting delay 207
LAN, setting for 100 Progress Indicator in Alerting 207
messaging server host 283 Progress Indicator in Call Proceeding 206
reserving in network server 99 Progress Indicator in Connect 207
SIP outbound proxy 283 Progress Indicator in Progress 207
SIP proxy 283 Progress Indicator in Setup 206
SIP registrar 283 Progress Indicator in Setup ACK 206
syslog daemon 71 select interface 206
unit, setting for 100 introduction 177
vocal identification of 161 IP synchronization 216
IP address call 434 enabling 217
IP routing master units 217
DHCPv4 classless static route option 131 resetting synchronization data 218
IPv4 forwarding 127 slave units 218
rule, creating 128 Layer 1 Timer 3 210
rule, deleting 129 preset 181
rule, editing 128 PRI interface
rule, moving 129 calling name max length 190
static IPv4 routes 130 channel allocation strategy 189
status information 129, 130 channel range 188
IP synchronization, in E&M 270 clock mode 186
enabling 270 copy config to all interfaces 185, 520
master units 270 encoding scheme, fallback 188
resetting synchronization data 271 encoding scheme, preferred 188
slave units 271 endpoint type 185
IP synchronization, in ISDN 216 exclusive B-channel selection 190
enabling 217 inband DTMF dialing 189
master units 217 inband tone generation 189
resetting synchronization data 218 InformationFollowing operation 194
slave units 218 line coding 187
IP synchronization, in R2 246 line framing 187
enabling 246 line type 180
master units 246 link establishment 190, 200
resetting synchronization data 247 maximum active calls 189
slave units 247 network location 187
IPv4 overlap dialing 190
forwarding 127 restart on startup 190
IPv4, static routes 130 select interface 185
ISDN sending complete information 190
partial reset 181, 597 signal information element 189
ISDN configuration signalling protocol 187
auto calcel timeout 210 statistics 181
auto-configuration 180 services
BRI interface AOC-E support 214, 215
bypass connection 205 CLIP 212
calling name max length 200 CLIR 212
channel allocation strategy 199 CLIR override 213
clock mode 197 COLP 213
connection type 197 COLR 213
copy config to all interfaces 196 COLR override 214
encoding scheme, fallback 198 facility services, enabling 211
encoding scheme, preferred 198 Maintenance Service Call Termination 214
endpoint type 196 MSN 216
exclusive B-channel selection 200 select interface 211
inband DTMF dialing 199 supplementary services, enabling 211
inband tone generation 199
maximum active calls 199
network location 198 J
overlap dialing 200 jitter buffer protection 387
restart on startup 200
select interface 196
sending complete information 200
K
signal information element 199 keep alive, in SIP 285

keys, CLI keyword 33 transformation 490

type
accessing 485
L input call property 486
Lan1, network interface 100 moving 494
Last Diverting E.164, routing type 469 transformation 486
Last Diverting Number Presentation, routing type 470 Max-Forwards header, in SIP 320
Last Diverting Party Number Type, routing type 469 maximum active calls
Last Diverting Private Type of Number, routing type 469 E&M interface, configuration 259
Last Diverting Public Type of Number, routing type 469 ISDN interface
Last Diverting Reason, routing type 468 BRI configuration 199
Layer 1 Timer 3, ISDN interface 210 PRI configuration 189
lease times in embedded DHCP server 151 R2 interface, configuration 227
leases in embedded DHCP server 155 maximum facility waiting delay, ISDN interface 207
LEDs behaviour message waiting indicator 437
in firmware pack update mode 556 visual type 439
licence, activate 599 messaging subscription, in SIP 349
limitations, DSP 6 MFR1
line coding dialling delay between 454
E&M interface, configuration 258 MSN, ISDN interface 216
ISDN interface MTU
PRI configuration 187 configuring 108
R2 interface, configuration 226 MTU, requirements 639
line framing Mu (µ)-Law 353
E&M interface, configuration 258 defined 715
ISDN interface music on hold
PRI configuration 187 configuring 447
R2 interface, configuration 226 remote server
line signalling protocol HTTP server, configuring 447
R2 interface, configuration 227 TFTP server, configuring 447
line type
PRI configuration 61, 180, 222, 254
link establishment, ISDN interface N
PRI configuration 190, 200 name, CLI keyword 35
link state verification timeout, on FXO port 171 NAT
link state verification, on FXO port 171 disabling 148
LLDP configuration, network interface 107 in configuration backup image 545
local ring behaviour, in SIP 324 in configuration script download 536
locking endpoints 67 partial reset 141
logs off, CLI keyword 34 NAT, destination
logs on, CLI keyword 34 rule, creating 145
loop current 166 rule, deleting 148
ls, CLI keyword 34 rule, editing 145
rule, moving 148
NAT, source
M rule, creating 141
MAC address 99 rule, deleting 148
defined 715 rule, editing 141
vocal identification of 161 rule, moving 148
machine detection NBNS server, in embedded DHCP server 155
CED tone detection 396 network interfaces
CED tone detection, behaviour 396 connection type 100
CNG tone detection 396 defining 100
V.21 modulation detection 396 DHCP client identifier presentation
Maintenance Service Call Termination, ISDN interface 214 DHCP
management interface client identifier presentation 106
partial reset 599 Ethernet link configuration 108
selecting 599 Lan1 100
mappings, in call routing LLDP configuration 107
defined 484 partial reset 99
expression PPP configuration 105
accessing 486 PPP negotiation 105
criteria 487 priority 104
deleting 494 Rescue 100
moving 494 Uplink 100
sub mappings 493 network location, ISDN interface

Index
BRI configuration 198 PCM capture 74

PRI configuration 187 penalty box, in SIP 333
network parameters ping, CLI keyword 36, 37
automatic configuration interface 89 port number
default gateway client 91 configuration backup server 541
DNS client 92 configuration script download server 531
host 89 firmware pack update files server 553
SNTP client 93 messaging serverr host 283
syslocation, updating 98 SIP outbound proxy 283
sysname, updating 98 SIP proxy 283
time zone, defining custom 94 SIP registrar 283
network traffic control POTS
in QoS 118 auto cancal timeout 163
not allowed behavior, on FXO port 170 caller ID customization 160
NTP server, in embedded DHCP server 154 FXO configuration
answering on caller ID 170
call failure timeout 172
O detection custom repetition 174
objectives of document xxiv dial tone detection mode 169
objects, CLI keyword 35 dial tone detection timeout 169
Offer/Answer model 312, 315, 319 force end of call on call failure 171
Original Diverting E.164, routing type 469 forced end of call on silence detection mode 172
Original Diverting Number Presentation, routing type 470 forced end of call on tone detection mode 172
Original Diverting Party Number Type, routing type 469 link state verification 171
Original Diverting Private Type of Number, routing type 469 link state verification timeout 171
Original Diverting Public Type of Number, routing type 469 not allowed behavior 170
Original Diverting Reason, routing type 468 pre dial delay 169
outbound proxy server, configuring 282 silence detection timeout 172
out-of-band DTMF 391 tone detection custom cadence 173
overlap dialing, ISDN interface tone detection custom frequency 173
BRI configuration 200 wait before answering delay 170
PRI configuration 190 wait foe callee to answer 170
FXS bypass
P activation 167
activation DTMF map 168
packetization time deactivation timeout 168
Clear Channel 368 FXS Country Override Flash Hook Detection Range 166
Clear Mode 367 FXS Country Override Loop Current 166
G.711 361 FXS Disconnect Delay 163
G.723 363 FXS Inband Ringback 164
G.726 365 FXS Line Supervision Mode 163
G.729 366 FXS Override Country Customization 165
X-CCD Clear Channel 370 FXS power drop on disconnect duration 164
partial reset 15, 18 FXS service activation 164
disabling 15 FXS shutdown behavior 164
E&M preset 256 status information 159
ISDN preset 181, 597 power feeding over ISDN 62
local firewall service 121 PPP configuration, network interface 105
management interface 599 PPP negotiation 105
NAT service 141 PPPoE connection type 100
network interfaces 99 PRACK, support 342
R2 preset 224 pre dial delay, on FXO port 169
RADIUS authentication 593 preset
SNMP 563, 568 E&M interfaces 255
user access 18 ISDN interfaces 181
user password 592 R2 interfaces 223
see also factory reset PRI interface configuration
password calling name max length 190
modify 597, 599 channel allocation strategy 189
payload type channel range 188
Clear Channel 368 clock mode 186
Clear Mode 367 clock reference 61
G.726 365 copy config to all interfaces 185, 520
using the one found in answer 394 encoding scheme, fallback 188
X-CCD Clear Channel 370 encoding scheme, preferred 188
pcapture, CLI keyword 36 endpoint type 185

exclusive B-channel selection 190 maximum active calls 227

inband DTMF dialing 189 preset 223
inband tone generation 189 PRI statistics 248
InformationFollowing operation 194 R2 digit timers 236
line coding 187 R2 link timers 237
line framing 187 R2 signaling 228
line type 61, 180, 222, 254 C bit setting 229
link establishment 190, 200 D bit setting 229
maximum active calls 189 override default country settings 229
network location 187 R2 timers 232
overlap dialing 190 R2 tones backward groups 242
restart on startup 190 R2 tones forward groups 241, 269
select interface 185 register signalling protocol 227
sending complete information 190 selecting R2 signaling protocol 222
signal information element 189 RADIUS
signaling 61, 222, 254 accounting type 593
signalling protocol 187 authentication type 592
priority, of network interfaces 104 servers access rights 594
privacy key servers configuration 593
configuration backup image 541 Reason header 341
configuration script download 531 Referred-by header 341
privacy level of calls 496 register home domain override, in SIP 326
Progress Indicator in Alerting, ISDN interface 207 register signalling protocol
Progress Indicator in Call Proceeding, ISDN interface 206 R2 interface 227
Progress Indicator in Connect, ISDN interface 207 registrar server, configuring 282
Progress Indicator in Progress, ISDN interface 207 registration
Progress Indicator in Setup ACK, ISDN interface 206 delay value, in SIP 299
Progress Indicator in Setup, ISDN interface 206 expiration value, in SIP 295
proxy server expiration, in SIP 292, 295
SNMP, configuring via refresh, in SIP 292, 294
override 326 regular expression, defined 463
proxy server, configuring 282 related documentation xxv
Require header, ignoring, in SIP 328
Rescue, network interface 100
Q reset
QoS factory reset procedure 16
defined 717 partial reset procedure 15
network traffic control 118 disabling 15
VLAN 117 resolve route header, in SIP 327
QoS, local restart
802.1q 117 unit 49, 53
Differentiated Services (DS) Field 115 restart on startup, ISDN interface
QSIG BRI configuration 200
signalling protocol 187, 198 PRI configuration 190
ring
R distinctive 605
ring management 65
R2 routes, in call routing
partial reset 224 call property to compare 481
R2 configuration creating 480
auto-configuration 223 defined 479
channel allocation strategy 226 deleting 484
channel range 226 destination, of call 483
clock mode 225 editing 480
country selection 227 expression to compare 482
encoding scheme 227 mapping, apply 483
encoding scheme, preferred 227 moving 484
introduction 221 signalling, of call 483
IP synchronization 246 source to compare 481
enabling 246 RTCP 115
master units 246 RTP
resetting synchronization data 247 enforcing symmetric 379
slave units 247 RTP statistics
line coding 226 collection period 383
line framing 226 end-of-connection notification 383
line signalling protocol 227 end-of-period notification 384

Index
statistics displayed 381 built from 500

RTP, early creating 499
listening to 327 defined 498
rule, in events deleting 501
deleting 79 editing 499
modified header 499
moving 501
S name 499
scalars, CLI keyword 37 SIP INFO
SDP call waiting private number for 320
direction attribute level 324 SIP INFO without content answer 331
SDP detect peer direction attribute, in SIP 322 SIP redirects, in call routing
SDP on hold connection address, in SIP 323 creating 513
second call, service 437 defined 513
secure communication deleting 514
introduction 50 editing 513
secure media 377 moving 514
secure signalling 307 SIP, setting
select interface 210 ACK branch matching 328
sending complete information, ISDN interface allow audio and image negotiation 316
BRI configuration 200 allow less media in response 316
PRI configuration 190 allow media reactivation in answer 316
services, ISDN authentication information
AOC-E support 214, 215 creating 302
CLIP 212 deleting 304
CLIR 212 editing 302
override 213 moving 304
COLP 213 call waiting private number for SIP INFO 320
COLR 213 calling party name of the caller ID 166
override 214 certificate trust level for TLS connections 306
enabling 211 default registration value in registrations 292
Maintenance Service Call Termination 214 default username value 312
MSN 216 direction attribute "sendonly" 317, 323
select interface 211 direction attributes 321
services, managing 53 direction attributes present 321
session ID/number in Origin field, in SIP 325 DNS SRV record lock 326
session refresh 343 early RTP, listening to 327
session timers 343 error causes mapping 334
set, CLI keyword 37 cause to SIP 338
show, CLI keyword 38 SIP to cause 336
signal information element, ISDN interface escape pound (#) in SIP URI Username 319
BRI configuration 199 event handling 348
PRI configuration 189 expiration value in registration 295
signaling force DNS NAPTR in TLS 309
PRI configuration 61, 222, 254 forked provisional reponses behaviour 343
signalling properties, in call routing gateways
180 with SDP 496 adding 277
183 without SDP 496 auto-routing, setting for 519
call properties translation 497 keep alive destinations 286
creating 495 messaging servers 284
defined 494 proxy servers 284
deleting 498 registrar servers 283
destination host 496 removing 277
early connect 495 home domain proxy override 345
early disconnect 496 Ignore OPTONS on no usable endpoints 313
editing 495 ignore plus (+) character in username 319
moving 498 ignore Require header 328
name 495 keep alive 285
privacy level 496 local ring behaviour 324
SIP headers translation 497 map plus to TON International 319
signalling protocol Max-Forwards header 320
ISDN interface messaging subscription 349
BRI configuration 198 Offer/Answer model 312, 315, 319
PRI configuration 187 on hold SDP connection address 323
silence detection timeout, on FXO port 172 OPTIONS Method Support 312
SIP headers translation, in call routing outbound proxy server 282

loose router status 287 SNTP

payload type in answer, using 394 client 93
penalty box 333 remote server, configuring 526, 541, 550
persistent TLS connection port 306 time zone, defining custom 93
persistent TLS retry interval 306 Spanning Tree Protocol 556
PRACK 342 IPv6 stateless autoconfiguration 103
proxy server 282 special tags, in routing types values 470
quantity of RTP packets 395 special vocal features
Reason header 341 IP address 161
Referred-by header 341 MAC address 161
register home domain override 326 SSH
registrar server 282 accessing CLI via 22
registration configuration 291 standards supported xxix
registration contact matching 314 Static connection type 100
registration delay value 299 statistics
registration expiration 292, 295 SNMP 568
registration refresh 292, 294 subnets
reject code for no resource 329 managing in embedded DHCP server 151
reject code for unsupported SDP offer 330 supplementary ISDN services
resolve route header 327 enabling 211
ringing response code 329 sysinfo, CLI keyword 39
SDP detect peer direction attribute 322 syslocation, updating 98
SDP direction attribute level 324 syslog daemon
secure header 312 configuring the application 75
session ID/number in Origin field 325 diagnostic traces 71
session refresh 343 PCM capture 74
session timers 343 IP address 71
SIP INFO without content answer 331 messages level 71
supported DNS queries 348 view messages in the CLI 25
symmetric RTP, enforcing 379 sysname, updating 98
TCP/TLS connect timeout 307, 346
TLS certificate validation 318
TLS client authentication 308 T
TLS persistent connections status 282 T.38
transmission timeout 314 defined 357
transport type 305 enabling 373
TCP 305 number of redundancy packets 373
UDP 305 priority 373
UDP source port behaviour 308 tables, CLI keyword 39
unit registration 291 TCP
unsupported Content-Type 332 connect timeout 307, 346
UPDATE 342 transport type 305
user agents Telnet
authentication information 301 accessing CLI via 21
contact domain 290 TFTP
friendly user name 289 remote server, configuring 447, 525, 541, 550
header, enabling to send 299 time zone, configuring 93, 94
main user name 289 TLS
User-Agent Header certificate trust levels for connections in 306
format 330 certificate validation 318
voiceband data mode, call in 390 client authentication 308
SipEp, service connect timeout 307, 346
endpoint behaviour when unregistered 296 force DNS NAPTR in 309
registration parameters 294 persistent connection port 306
user behaviour when unregistered 297 persistent connections status 282
SNMP persistent retry interval 306
command tone detection custom cadence, on FXO port 173
syntax, normal 630 tone detection custom frequency, on FXO port 173
syntax, row 631 TR-069 configuration 575
partial reset 563, 568 TR-104 configuration 576
SNMPv3 privacy 563 transfer protocols
statistics 568 configuration backup image 541
traps, sending 563 configuration script download 531
SNMP, service firmware pack update 553
sending traps 563 transferring a call
statistics 568 attended transfer 425

Index
blind 425 blind 424

transmission timeout, setting 314 call waiting 426
traps, where to send in SNMP 563 conference call 429
type, CLI keyword 39 introduction 41
second call 436
using 43
U Web, service
U 108 TCP port 41
UDP
source port behaviour 308
transport type 305 X
unalias, CLI keyword 28 X-CCD Clear Channel
unit defined 357
locking/unlocking endpoints 67 enabling
restarting 49, 53 for data transmission 370
shutting down behaviour 69 for voice transmission 370
unlocking endpoints 67 packetization time 370
UPDATE, support 342 payload type 370
Uplink, network interface 100 priority
user for data codecs 370
add 591 for voice codecs 370
delete 591
modify 591
unregistered, behaviour when 297
user access, setting 17, 18
partial reset 18
secure password policies 17
user gain
input 452
output 452
User-Agent Header
HTTP format 42
SIP format 330
using this manual xxix
V
V.21 modulation detection 396
VLAN
creating 113
in QoS 117
vocal features, special
IP address 161
MAC address 161
vocal unit information 161
voice activity detection
G.729 366
voiceband data mode
starting a call in 390
W
wait before answering delay, on FXO port 170
wait for calle to answer, on FXO port 170
web interface
automatic call 419
call completion
CCBS 420
CCNR 420
call forward
on busy 409
on no answer 412
unconditional 414
call hold 435
call transfer
attended 424

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TA7102i, Software Configuration Guide

Aastra TA7102i, Software Configuration Guide

Dgw v2.0 Application TA7102i, Software Configuration Guide

© Copyright Aastra Technologies Limited, 2014.

P About this Manual

This Software Configuration Guide is intended for the following users:

Dgw v2.0 Application iii

Table 1: Software Configuration Guide Chapter/Appendices

“Chapter 8 - Events Configuration” on page 63 Describes how to associate a NOTIFICATION

“Chapter 16 - IP Routing Configuration” on Describes how to configure the unit's IP routing

iv Dgw v2.0 Application

Table 1: Software Configuration Guide Chapter/Appendices (Continued)

Dgw v2.0 Application v

Table 1: Software Configuration Guide Chapter/Appendices (Continued)

Call Router Parameters

“Chapter 39 - Auto-Routing Configuration” on Describes the call router service.

“Chapter 39 - Auto-Routing Configuration” on Describes the auto-routing feature.

“Chapter 43 - Certificates Management” on Describes how to transfer and manage certificates

vi Dgw v2.0 Application

Where to find Translated Warning Definition

Note: Indicates important information about the current topic.

SCN vs. PSTN

Dgw v2.0 Application vii

viii Dgw v2.0 Application

IP connectivity for analog phones and faxes

Dgw v2.0 Application 1

Table 2: Aastra Units Key Features (Continued)

Offered Chanels Processing Capacity

2 Dgw v2.0 Application

Figure 1: Management Interfaces

Table 4: Management Options

Management Choice Description Features

Dgw v2.0 Application 3

Table 4: Management Options (Continued)

Management Choice Description Features

4 Dgw v2.0 Application

The RESET/DEFAULT button allows you to:

Dgw v2.0 Application 5

 To trigger the Partial Reset:

Disabling the Partial Reset

6 Dgw v2.0 Application

 The reset action restarts the unit.

 To change the partial reset behaviour:

 To trigger the Factory Reset:

Dgw v2.0 Application 7

 To configure the Aastra unit user access:

Secure Password Policies

8 Dgw v2.0 Application

Table 8: Secure Password Policies (Continued)

Where to Go From Here

Dgw v2.0 Application 9

10 Dgw v2.0 Application

2 Command Line Interface (CLI)

Configuring the CLI

 To configure the CLI access:

Dgw v2.0 Application 11

Accessing the CLI

You can access the CLI a Telnet or SSH session.

Accessing the CLI via a Telnet Session