Aastra Telecom will not accept liability for any damages and/or long distance charges, which result from unauthorized and/or
unlawful use. While every effort has been made to ensure accuracy, Aastra Telecom will not be liable for technical or editorial errors
or omissions contained within this documentation. The information contained in this documentation is subject to change without
notice.
General Information ..................................................................................................................................2
Release Content Information ................................................................................................................2
Before you Upgrade .................................................................................................................................. 3
Please Read Before Upgrading the Phones to Release 2.3 ................................................................. 3
New Features in Release 2.3 .................................................................................................................... 4
Issues Resolved in Release 2.3 ..............................................................................................................63
Contacting Aastra Telecom Support ...................................................................................................... 65
SIP Features ....................................................................................................................................... 57
P-Asserted-Identity (PAI) Support in UPDATE message .............................................................. 57
DTMF Tones in Info Requests ......................................................................................................57
Ignore Out of Sequence Errors .....................................................................................................58
SIP BLA Expires Timer .................................................................................................................59
Troubleshooting Features ...................................................................................................................61
This Release Note 2.3 provides new features for the 5i Series IP Phones (51i, 53i, 55i, 57i, and 57i
CT), and the 9143i, 9480i, and 9480i CT IP Phones. It also includes the issues resolved since
Release 2.2.1 of the IP Phones.
For more detailed information about features associated with each phone, and for information on
how to use the phones, see your model-specific SIP IP Phone Installation Guide and the SIP IP
Phone User Guide. For detailed information about more advanced features, see the SIP IP Phone
Administrator Guide.
SIP IP Phone
and 5i Series Phones
Release Note 2.3
Topics in this release note include:
•General Information
(release content, hardware supported, bootloader requirements)
•New Features in Release 2.3
•Issues Resolved in Release 2.3
•Contacting Aastra Telecom Support
RN-001029-02, Release 2.3, Rev 00 1
IP Phone Release Notes 2.3
General Information
General Information
Release Content Information
This document provides release content information on the Aastra 9143i, 9480i, 9480i CT and 5i
Series SIP IP phone firmware.
ModelRelease NameRelease VersionRelease FilenameRelease Date
51iGeneric SIP2.3.0.82FC-001126-02-REV00July 2008
53iGeneric SIP2.3.0.82FC-001086-04-REV00July 2008
55iGeneric SIP2.3.0.82FC-001087-03-REV00July 2008
57iGeneric SIP2.3.0.82FC-001088-03-REV00July 2008
57i CTGeneric SIP2.3.0.82FC-001089-03-REV00 July 2008
This release of firmware is compatible with the following Aastra IP portfolio products:
•51i
•53i
•55i
•57i
•57i CT
•9143i
•9480i
•9480i CT
Bootloader Requirements
This release of firmware is compatible with the following Aastra IP portfolio product bootloader
versions:
•51i - Bootloader 2.1.0.2088 or higher
•53i - Bootloader 2.0.1.1055 or higher
•55i - Bootloader 2.0.1.1055 or higher
•57i - Bootloader 2.0.1.1055 or higher
•57i CT - Bootloader 2.0.1.1055 or higher
•9143i - Bootloader 2.2.0.166 or higher
•9480i - Bootloader 2.2.0.166 or higher
•9480i CT - Bootloader 2.2.0.166 or higher
2RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
Before you Upgrade
Please Read Before Upgrading the Phones to Release 2.3
Release 2.3 provides several new features for the Aastra IP Phones. However, before upgrading
the phones in your network to Release 2.3, please read the following IMPORTANT information:
•In Release 2.3, LLDP is enabled by default.
If LLDP is enabled on your network, the phones may come up with different network settings.
For more information about LLDP, see “Link Layer Discovery Protocol for Media Endpoint
Devices (LLDP-MED)” on page 7.
•Release 2.3 includes support for DHCP Options 159 and 160.
If the DHCP server supplies Options 159 and 160, the phones will attempt to contact the
configuration server given in these options. For more information about Options 159 and 160,
see “DHCP Options 159 and 160 for the Configuration Server” on page 12.
•Release 2.3 provides HTTPS validation.
If you are using HTTPS and the certificates are not valid or are not signed by Verisign,
Thawte, or GeoTrust, the phones fail to download configuration files. For more information
about HTTPS validation, see “HTTPS Server Certificate Validation” on page 23.
Before you Upgrade
•Release 2.3 includes a Watchdog task feature.
If the phone detects a failure (for example, a crash), the phone automatically reboots. For more
information about Options 159 and 160, see “WatchDog Task Feature” on page 61.
Note: If you factory default a phone with Release 2.3 software, when the
phone reboots, it attempts to connect to rcs.aastra.com. There is no
personal information transmitted from the phone and the phone continues
to boot up as normal.
RN-001029-02, Release 2.3, Rev 003
IP Phone Release Notes 2.3
New Features in Release 2.3
New Features in Release 2.3
Description
This section provides the new features in SIP IP Phone Release 2.3. These new features apply to
all of the 5i Series phones and the 9143i, 9480i, and 9480i CT phones, unless specifically stated
otherwise. Each feature also specifies whether it affects the Administrator, the User, or both.
FeatureDescription
Network Features
Link Layer Discovery Protocol for Media
Endpoint Devices (LLDP-MED)
(For Administrator)
DHCP Options 159 and 160 for the
Configuration Server
(For Administrator)
DHCP Option 12 Hostname for the
Configuration Server
(For Administrator)
DHCP Option 77 User Class for the
Configuration Server
(For Administrator)
Multiple DHCP Servers
(For Administrator)
Security Features
Configuration File Encryption
(For Administrator)
HTTPS Server Certificate Validation
(For Administrator)
XML Features
XML Execute Commands for Playing a WAV
File
The IP Phones support Link Layer Discovery Protocol
for Media Endpoint Devices (LLDP-MED).
In addition to DHCP options 43 and 66 already
supported on the IP Phones, Release 2.3 provides new
DHCP Options 159 and 160.
The IP Phones support DHCP Option 12 that the phone
automatically sends to the configuration server. This
option specifies the hostname (name of the client).
The IP Phones support DHCP Option 77 User Class,
that is sent in DHCP request packets from the phone to
the configuration server. This Option 77 defines specific
User Class identifiers to convey information about a
phone’s software configuration or about its user's
preferences.
In Release 2.3, the IP Phones can now receive
messages from multiple DHCP servers.
Some vendors can have specific methods to encrypt
files on their configuration servers. For each phone, the
configuration server can generate a random hex string
(encryption key) that is used to encrypt the phone’s
MAC-specific configuration file.
The HTTPS client on the IP Phones now support the
validation of HTTPS certificates.
The IP Phones now allow a WAV file to be played or
stopped via XML Execute commands.
(For XML Developers)
RTP Recording and Simultaneous Playing
(not supported on 51i)
(For XML Developers)
4RN-001029-02, Release 2.3, Rev 00
The IP Phones allow for RTP recording and
simultaneous playing of an audio file via XML Execute
commands.
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
FeatureDescription
New Features in Release 2.3
Dialpad Passthrough for Objects
(For XML Developers)
Non-Blocking Action URI
(For Administrator)
New XML URI Variables
(For Administrator and XML Developers)
XML Web Applications Button
(For Administrator and User)
XML Key Redirection
(For XML Developers)
Options Key Redirection (for Options Menu
on all phones and Services Menu on 51i)
(For XML Developers)
XML Applications and Off-Hook Interaction
(For Administrator)
XML URI for Key Press Simulation
(For XML Developers)
Using XML Commands to Reset Local Data
on the Phone
(For XML Developers)
Action URI Disconnected Feature
(For Administrator)
User Interface Features
Preferred Line Focus Feature
(For Administrator & User)
On the IP Phones, an XML Developer can now control
digit passthrough while the phone is in the connected
state using the “allowDTMF” attribute with the XML
objects AastraIPPhoneTextScreen,
AastraIPPhoneFormattedTextScreen, and
AastraIPPhoneImageScreen.
The Action URI feature on the IP Phones is now
enhanced to prevent the phone from hanging if the
Action URIs should fail. Also, support for transparent,
non-blocking, XML post execute item URI was added.
Action URIs on the phone support variables, which are
replaced with their corresponding value when the URI is
triggered. New XML variables have been added that can
be used with an onhook Action URI, and allow for
enhanced information in call records and billing
applications:
The IP Phones now have a feature that allows a User or
Administrator to access XML-based applications via a
new pre-programmed softkey called, "WebApps".
The IP phones now allow the redirecting of phone-based
hard keys to XML scripts.
The IP phones now allow the redirecting of the Options
Key (Options Menu on all phones and Services Menu on
the 51i) to an XML script.
In Release 2.3, a feature has been implemented that
prevents the phone from going into the off-hook/dialing
state when the handset is off-hook and the call ends.
Release 2.3 provides a feature that allows you to define
XML Key URIs that can send key press events to the
phone, just as if the physical hard key or softkey were
pressed on the phone.
Release 2.3 provides new XML commands that allow
you to delete and reset the phone’s directory, callers list,
redial list, and the local.cfg file. You use these
commands with the AastraIPPhoneExecute object.
A new parameter, “action uri disconnected” has been
added to Release 2.3, that allows a disconnect event to
occur when the phone transitions from any active state
(outgoing, incoming, connected, or calling) to an idle
state.
In previous releases of the IP Phones, after a call
(incoming or outgoing), the phone would stay on the line
that was used for that call. In Release 2.3, an
Administrator or User can now define a preferred line as
well as a preferred timeout.
RN-001029-02, Release 2.3, Rev 005
IP Phone Release Notes 2.3
New Features in Release 2.3
FeatureDescription
Dialpad Speeddial Supported on All Phones
(For Administrator & User)
UTF- 8 Codec for Multi-National Language
Support
(Applies to Administrator & User UIs)
Addition of New Timezone and Country
Codes
(Administrator & User)
SIP Features
P-Asserted-Identity (PAI) Support in UPDATE
message
(For Administrator)
DTMF Tones in Info Requests
(For Administrator)
Ignore Out of Sequence Errors
(For Administrator)
SIP BLA Expires Timer
(For Administrator)
Troubleshooting Features
WatchDog Task Feature
(For Administrator)
Previously, only the 51i IP Phone supported the dialpad
speeddial feature. In Release 2.3, all the phones now
support this feature. Using the IP Phone UI or the Aastra
Web UI, an Administrator or User can create speeddial
keys on the dialpad.
The IP Phones, expansion modules, and cordless
handsets previously supported ISO 8859-1 (Latin1)
language. The IP Phones and expansion modules now
include support for ISO 8859-2 (Latin2) of multi-national
languages when displaying and inputting in the IP Phone
UI and the Aastra Web UI.
Note: UTF-8 is not applicable to the Handsets for CT
models.
Five new timezone and country codes have been added
to the 2.3 Release.
In Release 2.3, the phones now support PAI header in
the UPDATE message, according to
draft-ietf-sipping-update-pai-00.
In Release 2.3, the phones now support decoding and
playing out DTMF tones sent in SIP INFO requests.
In Release 2.3, you can configure the phone via the “sip accept out of order requests” parameter to ignore
CSeq number errors on all SIP dialogs on the phone.
Release 2.3 now includes a SIP BLA subscription period parameter that allows an Administrator to set the
BLA expiration timer for how long the phone waits to
receive a BLA subscribe message from the server.
Release 2.3 provides a troubleshooting feature called
the “WatchDog” task that monitors the status of the
phones and provides the ability to get stack traces from
the last time the phone failed. When the phone detects a
failure (i.e., a crash), it automatically reboots.
The following paragraphs describe these features in more detail.
6RN-001029-02, Release 2.3, Rev 00
Network Features
Link Layer Discovery Protocol for Media Endpoint Devices (LLDP-MED)
In Release 2.3, the IP Phones support Link Layer Discovery Protocol for Media Endpoint Devices
(LLDP-MED). LLDP-MED is designed to allow for things such as:
•Auto-discovery of LAN policies (such as VLAN, Layer 2 Priority and Diffserv settings)
leading to "plug and play" networking.
•Extended and automated power management of Power over Ethernet endpoints.
•Inventory management, allowing network administrators to track their network devices, and
determine their characteristics (manufacturer, software and hardware versions, serial / asset
number).
On the IP Phones, LLDP-MED performs the following:
•Supports the MAC/PHY configuration (e.g. speed rate/duplex mode).
•Supports VLAN info from the network policy; this takes precedence over manual settings.
•Allows you to enable/disable LLDP-MED if required.
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
•Allows you to configure time interval between successive LLDP Data Unit (LLDPDU)
frames.
•Allows LLDP packets to be received from the LAN port and allows them to reach the PC Port.
•Allows the phone to use the location information, Explicit Congestion Notification (ECN)
Emergency Location Identification Number (ELIN), sent by the switch, as a caller ID for
making emergency calls.
Note: If the phone receives location information in ECN ELIN format
(10 to 25 numeric string), the phone replaces the caller ID SIP header
with the ECN ELIN value and the SIP URI does not change. The phone
determines if this is an emergency number by checking the emergency
dial plan configured on the phone.
You can enable or disable the LLDP-MED on the IP Phones using the configuration files, the IP
Phone UI or the Aastra Web UI.
RN-001029-02, Release 2.3, Rev 007
IP Phone Release Notes 2.3
New Features in Release 2.3
Configuring LLDP-MED Using the Configuration Files
Use the following parameters to configure LLDP-MED on the IP Phones.
Parameter–
lldp
LLDP
(in Web UI)
DescriptionEnables or disables Link Layer Discovery Protocol for Media Endpoint
FormatBoolean
Default Value1 (enabled)
Range0 (disabled)
Examplelldp: 0
Parameter–
lldp interval
LLDP Packet Interval
(in Web UI)
DescriptionThe amount of time, in seconds, between the transmission of LLDP Data
FormatInteger
Default Value30
Range0 to 2147483647
Examplelldp interval: 60
Configuration Filesaastra.cfg, <mac>.cfg
IP Phone UIOptions->Administrator Menu->
Network Settings->Ethernet&VLAN->
LLDP Support
Aastra Web UIAdvanced Settings->Network->
Advanced Network Settings
Devices (LLDP-MED) on the IP Phone.
1 (enabled)
Configuration Filesaastra.cfg, <mac>.cfg
Aastra Web UIAdvanced Settings->Network->
Advanced Network Settings
Unit (LLDPDU) packets. The value of zero (0) disables this parameter.
Parameter–
use lldp elin
Use LLDP ELIN
(in Web UI)
DescriptionEnables or disables the use of an Emergency Location Identification Number
FormatBoolean
Default Value1 (enabled)
Range0 (disabled)
Exampleuse lldp elin: 0
8RN-001029-02, Release 2.3, Rev 00
Configuration Filesaastra.cfg, <mac>.cfg
Aastra Web UIBasic Settings->Preferences->General
(ELIN) received from LLDP as a caller ID for emergency numbers.
1 (enabled)
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
Configuring LLDP-MED Using the IP Phone UI
Use the following procedure to enable/disable LLDP-MED using the IP Phone UI.
Note: You cannot configure the “LLDP Interval” or the “Use LLDP
ELIN” parameters via the IP Phone UI.
Aastra IP Phone UI
StepAction
1Press Options, and then select Administrator Menu.
On the 51i:
Press Services, and then select Options->Administrator Menu.
2Select Network Settings.
3Select Ethernet & VLAN.
New Features in Release 2.3
4Select LLDP Support.
5Press CHANGE to toggle the LLDP setting to Enabled or Disabled.
This field enables or disables Link Layer Discovery Protocol for Media Endpoint Devices (LLDP-MED) on the IP
Phone.
6Press DONE to save the change.
RN-001029-02, Release 2.3, Rev 009
IP Phone Release Notes 2.3
New Features in Release 2.3
Configuring LLDP-MED Using the Aastra Web UI
Use the following procedure to configure LLDP-MED using the Aastra Web UI:
Aastra Web UI
1Click on Advanced Settings->Network->Advanced Network Settings.
LLDP
LLDP Packet Interval
2The “LLDP” field is enabled by default. To disable LLDP, click the check mark in the box to clear the check mark.
3In the “LLDP Packet Interval” field, enter the time, in seconds, between the transmission of LLDP Data Unit
(LLDPDU) packets.
The value of zero (0) disables this parameter. Valid values are 0 to 2147483647. Default is 30.
4Click to save your changes.
10RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
Aastra Web UI
5Select Basic Settings->Preferences->General.
Use LLDP ELIN
New Features in Release 2.3
6The “Use LLDP ELIN” field is enabled by default. To disable LLDP ELIN, click the check mark in the box to clear
the check mark.
This field enables or disables the use of an Emergency Location Identification Number (ELIN) received from LLDP
as a caller ID for emergency numbers.
7Click to save your changes.
RN-001029-02, Release 2.3, Rev 0011
IP Phone Release Notes 2.3
New Features in Release 2.3
DHCP Options 159 and 160 for the Configuration Server
In addition to DHCP options 43 and 66 already supports on the IP Phones for setting the
configuration server, Release 2.3 provides new DHCP Options 159 and 160. The IP Phones now
use the following order of precedence when deriving the configuration server parameters: 43,
160, 159, 66.
In addition, an administrator can override this order of precedence by setting a configuration
parameter called, dhcp config option override. You can set this parameter via the configuration
files, IP Phone UI, or the Aastra Web UI. Setting this parameter results in the phone only using
the chosen DHCP option and ignoring the other options
Warning: Administrators should review the updated IP phone DHCP
option precedence order and configuration options to avoid potential
impact to existing Aastra IP phone deployments.
Configuring DHCP Option Override via the Configuration Files
Use the following parameter to configure DHCP option override on the IP Phones.
.
Parameter–
dhcp config option
override
DHCP Option Override
(in Web UI)
DescriptionThe value specified for this parameter overrides the precedence order for
FormatInteger
Default Value0 (None - no override - uses normal precedence order of
Range0
Exampledhcp config option override: 66
Configuration Filesaastra.cfg, <mac>.cfg
IP Phone UIOptions->Administrator Menu->
Network Settings->DHCP Settings->
Option Override
Aastra Web UIAdvanced Settings->Network->
Advanced Network Settings
determining a configuration server.
Note: You must restart the IP Phone for this parameter to take affect.
43, 160, 159, 66)
43
66
159
160
12RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
Configuring DHCP Option Override via the IP Phone UI
Use the following procedure to configure DHCP option override via the IP Phone UI.
Aastra IP Phone UI
StepAction
1Press Options, and then select Administrator Menu.
On the 51i:
Press Services, and then select Options->Administrator Menu.
2Select Network Settings.
3Select DHCP Settings.
4Select Option Override. The following list displays:
New Features in Release 2.3
•None (default) - no override - uses normal precedence order of
•Option 43
•Option 66
•Option 159
•Option 160
5Choose an option that you want to use to override the DHCP normal precedence order, and press DONE.
6Restart the phone for the selection to take affect.
43, 160, 159, 66.
RN-001029-02, Release 2.3, Rev 0013
IP Phone Release Notes 2.3
New Features in Release 2.3
Configuring DHCP Option Override via the Aastra Web UI
Use the following procedure to configure DHCP option override using the Aastra Web UI:
Aastra Web UI
1Click on Advanced Settings->Network->Advanced Network Settings.
DHCP Options
2In the “DHCP Option Override” field, select an option to use to override the normal precedence order. Valid values
are:
•None (default) - no override - uses normal precedence order of
•Option 43
•Option 66
•Option 159
•Option 160
3Click to save your changes.
4Click on Operation->Reset, and restart the phone for the changes to take affect.
14RN-001029-02, Release 2.3, Rev 00
43, 160, 159, 66.
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
DHCP Option 12 Hostname for the Configuration Server
Release 2.3 includes a new DHCP Option 12 that the phone automatically sends to the
configuration server. This option specifies the hostname (name of the client). The name may or
may not be qualified with the local domain name (based on RFC 2132). See RFC 1035 for
character set restrictions.
Notes:
1.The hostname of [<model><MAC address>] automatically
populates the field on initial startup of the phone. For example, for a 53i,
the “Hostname” field is automatically populated as “53i00085D164435”,
where the model number is “53i” and the MAC address is
“00085D164435”.
2.If the configuration server sends the hostname back to the phone
in a DHCP Reply Packet, the hostname is ignored.
An Administrator can change the “Hostname” for the DHCP Option 12 via the configuration files,
the IP Phone UI, and the Aastra Web UI.
Changing DHCP Option 12 Hostname via the Configuration Files
Use the following parameter to change the “hostname” for DHCP Option 12.
Parameter–
hostname
Hostname
(in Web UI)
DescriptionSpecifies the hostname DHCP Option 12 that the phone sends with the
FormatString
Default Value[<model><MAC IP Address>]
RangeUp to 64 alpha-numeric characters
Examplehostname: aastra4
Configuration Filesaastra.cfg, <mac>.cfg
IP Phone UIOptions->Administrator Menu->
Network Settings->Hostname
Aastra Web UIAdvanced Settings->Network->
Basic Network Settings
DHCP Request packet.
Note: If you change this parameter, you must restart your phone for the
change to take affect.
Note: The value for this parameter can also be a fully qualified domain
name.
RN-001029-02, Release 2.3, Rev 0015
IP Phone Release Notes 2.3
New Features in Release 2.3
Changing DHCP Option 12 Hostname via the IP Phone UI
Use the following procedure to change the “hostname” for DHCP Option 12.
Aastra IP Phone UI
StepAction
1Press Options, and then select Administrator Menu.
On the 51i:
Press Services, and then select Options->Administrator Menu.
2Select Network Settings.
3Select Hostname.
4By default, the “Hostname” field is automatically populated with [<Model><MAC address>] of your phone (for
example, 53i00085D164435).
If you want to change the hostname, enter a hostname for your phone in the “Hostname” field, then press DONE.
Valid values are up to 64 alpha-numeric characters. You can use a fully qualified domain name if required.
5Restart the phone for the change to take affect.
16RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
Changing DHCP Option 12 Hostname via the Aastra Web UI
Use the following procedure to change the “hostname” for DHCP Option 12.
Aastra Web UI
1Click on Advanced Settings->Network->Basic Network Settings.
Hostname
New Features in Release 2.3
2By default, the “Hostname” field is automatically populated with [<Model><MAC address>] of your phone (for
example, 53i00085D164435).
If you want to change the hostname, enter a hostname for your phone in the “Hostname” field.
Valid values are up to 64 alpha-numeric characters. You can use a fully qualified domain name if required.
3Click to save your changes.
Note: Changing the “Hostname” field requires a reboot of your phone.
4Click on Operation->Reset, and click RESTART.
RN-001029-02, Release 2.3, Rev 0017
IP Phone Release Notes 2.3
New Features in Release 2.3
DHCP Option 77 User Class for the Configuration Server
Release 2.3 includes a new DHCP Option 77 User Class, that is sent in DHCP request packets
from the phone to the configuration server. This Option 77 defines specific User Class identifiers
to convey information about a phone’s software configuration or about its user's preferences. For
example, you can use the User Class option to configure all phones in the Accounting
Department with different user preferences than the phones in the Marketing Department. A
DHCP server uses the User Class option to choose the address pool for which it allocates an
address from, and/or to select any other configuration option.
Notes:
1.If the User Class is not specified (left blank) in the DHCP request
packet, the phone does not send the User Class DHCP Option 77.
2.Multiple User Classes inside a DHCP Option 77 are not supported.
3.DHCP Option 77 may affect the precedence of DHCP Options,
dependent on the DHCP Server.
An Administrator can configure the DHCP Option 77 User Class via the configuration files, the
IP Phone UI, and the Aastra Web UI.
Configuring DHCP Option 77 User Class via the Configuration Files
Use the following parameter to configure the phone to use DHCP Option 77.
Parameter–
dhcp userclass
DHCP User Class
(in Web UI)
DescriptionSpecifies the User Class DHCP Option 77 that the phone sends to the
FormatString
Default Value““
RangeUp to 64 alpha-numeric characters
Exampledhcp userclass: admin
Configuration Filesaastra.cfg, <mac>.cfg
IP Phone UIOptions->Administrator Menu->
Network Settings->DHCP Settings->
DHCP User Class
Aastra Web UIAdvanced Settings->Network->
Advanced Network Settings
configuration server with the DHCP Request packet.
Note: If you specify a value for this parameter, you must restart your phone
for the change to take affect. Any change in its value during start-up results
in an automatic reboot.
18RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
Configuring DHCP Option 77 User Class via the IP Phone UI
Use the following procedure to configure the phone to use DHCP Option 77.
Aastra IP Phone UI
StepAction
1Press Options, and then select Administrator Menu.
On the 51i:
Press Services, and then select Options->Administrator Menu.
2Select Network Settings.
3Select DHCP Settings.
4Select DHCP User Class.
5In the “DHCP User Class” field, enter a DHCP User Class to apply to your phones, then press DONE.
Valid values are up to 64 alpha-numeric characters. For example, “admin”.
6Restart the phone for the change to take affect.
RN-001029-02, Release 2.3, Rev 0019
IP Phone Release Notes 2.3
New Features in Release 2.3
Configuring DHCP Option 77 User Class via the Aastra Web UI
Use the following procedure to configure the phone to use DHCP Option 77.
Aastra Web UI
1Click on Advanced Settings->Network->Advanced Network Settings.
DHCP User Class
2In the “DHCP User Class” field, enter a DHCP User Class to apply to your phones. For example, “admin”.
Valid values are up to 64 alpha-numeric characters.
3Click to save your changes.
Note: Entering a value in the “DHCP User Class” field requires a reboot of your phone.
4Click on Operation->Reset, and click RESTART.
20RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
Multiple DHCP Servers
In Release 2.3, the IP Phones can now receive messages from multiple DHCP servers.
After the phone receives its first DHCP message, it listens for a specific time period, for more
DHCP messages. If the first DHCP offer contains configuration server information (Options 43,
66, 159 or 160), then the phone times out and continues using the first DHCP offer, without
listening for more DHCP offers. If the first DHCP message contains no configuration server
information, the phone continues to listen for other DHCP messages. If the second DHCP message
contains configuration server information and other conditions, the phone chooses the second
DHCP message over the initial DHCP message.
Note: If the DHCP Option Override parameter is enabled with a value
(Option 43, Option 66, Option 159, or Option 160), the phone checks the
override option setting before timing out.
IMPORTANT NOTE
Users currently using multiple DHCP servers on a single network could be affected by this
new feature.
RN-001029-02, Release 2.3, Rev 0021
IP Phone Release Notes 2.3
New Features in Release 2.3
Security Features
Configuration File Encryption
Some vendors can have specific methods to encrypt files on their configuration servers. For each
phone, the configuration server can generate a random hex string (encryption key) that is used to
encrypt the phone’s MAC-specific configuration file.
The encryption key is placed in a plain text MAC-specific configuration file that the server
downloads to the phone. After the phone receives the file, it updates the encryption key.
This method of encryption does not affect the implementation of the Aastra method of file
encryption.
Note: The aastra.cfg file is not encrypted with this feature.
You can set the phone-specific encryption key using the configuration files only.
For more information about configuration file encryption, contact Aastra Technical Support.
Configuring Configuration File Encryption
Use the following parameter to configure configuration file encryption on the IP Phones.
Parameter–
config encryption key
DescriptionSpecifies the phone-specific encryption key that the configuration server
FormatString
Default ValueNot applicable
RangeString length of 4 to 32 alphanumeric characters
Exampleconfig encryption key: 123abcd
Configuration Filesaastra.cfg, <mac>.cfg
uses to encrypt in a MAC-specific configuration file.
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SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
HTTPS Server Certificate Validation
The HTTPS client on the IP Phones now support validation of HTTPS certificates. This new
feature supports the following:
•Verisign, GeoTrust, and Thawte signed certificates
•User-provided certificates
•Checking of hostnames
•Checking of certificate expiration
•Ability to disable any or all of the validation steps
•Phone displays a message when a certificate is rejected (except on check-sync operations)
All validation options are enabled by default.
Certificate Management
Aastra Provided Certificates
The phone comes with root certificates from Verisign, GeoTrust, and Thawte pre-loaded.
User Provided Certificates
The administrator has the option to upload their own certificates onto the phone. The phone
downloads these certificates in a file of .PEM format during boot time after configuration
downloads. The user-provided certificates are saved on the phone between firmware upgrades but
are deleted during a factory default. The download of the User-provided certificates are based on a
filename specified in the configuration parameter, https user certificates (Trusted Certificates Filename in the Aastra Web UI; User-provided certificates are not configurable via the IP Phone
UI).
Note: Certificates that are signed by providers other than Verisign,
GeoTrust or Thwate do not verify on the phone by default. The user can
overcome this by adding the root certificate of their certificate provider to
the use-provided certificate .PEM file.
RN-001029-02, Release 2.3, Rev 0023
IP Phone Release Notes 2.3
New Features in Release 2.3
Certificate Validation
Certificate validation is enabled by default. Validation occurs by checking that the certificates
are well formed and signed by one of the certificates in the trusted certificate set. It then checks
the expiration date on the certificate, and finally, compares the name in the certificate with the
address for which it was connected.
If any of these validation steps fail, the connection is rejected. Certificate validation is controlled
by three parameters which you can configure via the configuration files, the IP Phone UI, or the
Aastra Web UI:
Aastra Web UIAdvanced Settings->Network->HTTPS Settings
When this parameter is set to 1, the HTTPS client performs validation on
SSL certificates before accepting them.
Note: If you are using HTTPS as a configuration method, and use a self
signed certificate, you must set this parameter to “0” (disabled) before
upgrading to Release 2.3 of the IP Phones.
1 (enabled)
24RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
Parameter–
https validate expires
Check Certificate
Expiration
(in Web UI)
DescriptionEnables or disables the HTTPS validation of the expiration of the certificates.
FormatBoolean
Default Value1 (enabled)
Range0 (disabled)
Examplehttps validate expires: 0
Parameter–
https validate hostnames
Check Certificate
Hostnames
(in Web UI)
DescriptionEnables or disables the HTTPS validation of hostnames on the phone.
FormatBoolean
Default Value1 (enabled)
Range0 (disabled)
Examplehttps validate hostnames: 0
Configuration Filesaastra.cfg, <mac>.cfg
IP Phone UIOptions->Administrator Menu->
Aastra Web UIAdvanced Settings->Network->HTTPS Settings
1 (enabled)
Parameter–
https user certificates
Trusted Certificates
Filename
(in Web UI)
DescriptionSpecifies a file name for a .PEM file located on the configuration server. This
FormatAlphanumeric string in the format <filename.pem>
Default ValueNot applicable
RangeNot applicable
Examplehttps user certificates: trustedCerts.pem
RN-001029-02, Release 2.3, Rev 0025
Configuration Filesaastra.cfg, <mac>.cfg
Aastra Web UIAdvanced Settings->Network->HTTPS Settings
file contains the User-provided certificates in PEM format. These certificates
are used to validate peer certificates.
Note: You must disable the “https validate certificates” parameter in order
for the phone to accept the User-provided certificates.
IP Phone Release Notes 2.3
New Features in Release 2.3
Configuring HTTPS Server Certificate Validation via the IP Phone UI
Use the following procedure to configure HTTPS server certificate validation for the IP Phones
using the IP Phone UI.
Aastra IP Phone UI
StepAction
1Press Options, and then select Administrator Menu.
On the 51i:
Press Services, and then select Options->Administrator Menu.
2Select Configuration Server.
3Select HTTPS Settings->Cert. Validation.
The following list displays:
•Enable
•Check Expires
•Check Hostnames
Enable/Disable HTTPS Server Certificate Validation
4Select Enable.
5Press Change to toggle the “Enable” field to “Yes” or “No”.
Note: If you are using HTTPS as a configuration method, and use a self signed certificate, you must set this field to
“No” before upgrading to Release 2.3 of the IP Phones.
6Press DONE to save the change and return to the Certificates screen.
Note: This change is immediately applied after pressing DONE.
8Press Change to toggle the “Check Expires” field to “Yes” or “No”.
Notes:
1. This change is immediately applied after pressing DONE.
2. If the “Check Expires” parameter is set to Yes, the clock on the phone must be set for the phone to accept the
certificates.
9Press DONE to save the change and return to the Certificates screen.
Note: This change is immediately applied after pressing DONE.
Enable/Disable HTTPS Validate Hostnames
10Select Check Hostnames.
11Press Change to toggle the “Check Hostnames” field to “Yes” or “No”.
26RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
Aastra IP Phone UI
StepAction
12Press DONE to save the change and return to the Certificates screen.
Note: This change is immediately applied after pressing DONE.
13Press to exit the Options Menu and return to the idle screen.
Configuring HTTPS Server Certificate Validation via the
Aastra Web UI
Use the following procedure to configure HTTPS server certificate validation for the IP Phones
using the Aastra Web UI.
Aastra Web UI
1Click on Advanced Settings->Network->HTTPS Settings.
New Features in Release 2.3
HTTPS Validation
Certificate
Parameters
2The “Validate Certificates” field is enabled by default. To disable validation of certificates, click the check mark
in the box to clear the check mark.
When this parameter is enabled, the HTTPS client performs validation on SSL certificates before accepting them.
Notes:
1. This parameter is immediately applied after clicking the SAVE SETTINGS button.
2. If you are using HTTPS as a configuration method, and use a self signed certificate, you must disable
(uncheck) this field before upgrading to Release 2.3 of the IP Phones.
3The “Check Certificate Expiration” field is enabled by default. To disable validation of certificate expiration, click
the check mark in the box to clear the check mark.
When this parameter is enabled, the HTTPS client verifies whether or not a certificate has expired prior to
accepting the certificate.
Notes:
1. This parameter is immediately applied after clicking the SAVE SETTINGS button.
2. If the “Check Certificates Expiration” parameter is set to Yes, the clock on the phone must be set for the
phone to accept the certificates.
RN-001029-02, Release 2.3, Rev 0027
IP Phone Release Notes 2.3
New Features in Release 2.3
Aastra Web UI
4The “Check Certificate Hostnames” field is enabled by default. To disable validation of hostnames, click the
check mark in the box to clear the check mark.
Note: This parameter is immediately applied after clicking the SAVE SETTINGS button.
5If you require the download of User-provided certificates in a .PEM formatted file, enter the file name in the format
<filename.pem> in the “Trusted Certificates Filename” field. For example:
trustedCerts.pem
This parameter specifies a file name for a .PEM file located on the configuration server. This file contains the
User-provided certificates in PEM format. These certificates are used to validate peer certificates.
Notes:
1. You must disable the “Validate Certificates” field in order for the phone to accept the User-provided
certificates.
2. This parameter requires you restart the phone in order for it to take affect.
6Click to save your changes.
7If you entered a filename in the “Trusted Certificates Filename” field, click on Operation->Reset, and restart
the phone for the changes to take affect.
28RN-001029-02, Release 2.3, Rev 00
XML Features
XML Execute Commands for Playing a WAV File
The IP Phones now allow a WAV file to be played or stopped via XML Execute commands. A
WAV file is an audio file format standard for storing an audio bit stream on a system in raw,
uncompressed format or compressed format to reduce the file size.
A WAV file can be streamed to the phone using the HTTP protocol.
The WAV feature supports the following:
•Streaming of the WAV file to allow it to be locally played
•Allows you to abort the audio streaming by pressing the Goodbye key on the phone.
•Supports the HTTP file download protocol
•Supports the aLaw and uLaw codecs
•Calls “Action URIINCOMING” when the audio WAV starts, and
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
“Action URI ONHOOK” when the WAV file playing has completed. The name of the WAV
file is inserted into the Action URI.
•Plays the WAV file only if the phone is idle.
•Aborts streaming if another line is selected for dialing out, or if the current line is accepting an
incoming call.
•WAV audio file starts only when 4 seconds of the audio data (or the complete file) has been
accumulated.
•Follows the standard phone behavior of speaker/head set/hand set.
An Administrator can use two new XML Execute commands with the AastraIPPhoneExecute
object to allow the phones to play a WAV file:
•Wav.Play - This command initiates the streaming of a WAV file to the phone.
•Wav.Stop - This command aborts a WAV streaming.
RN-001029-02, Release 2.3, Rev 0029
IP Phone Release Notes 2.3
New Features in Release 2.3
XML Command: Wav.Play
The Wav.Play XML Execute command starts the streaming of a WAV file. You enter this
command with the AastraIPPhoneExecute object in the following format:
The Wav.Stop Execute command aborts a WAV streaming currently in progress. You enter this
command with the AastraIPPhoneExecute object in the following format:
Wav.Stop:
Example
<AastraIPPhoneExecute>
<ExecuteItem URI="Wav.Stop:" />
<AastraIPPhoneExecute>
IP Phone UI Screens During WAV Streaming
The 9143i, 51i, and 53i LCD screens display the following during WAV streaming.
Streaming
00:10
The 9480i, 9480i CT, 55i, 57i, and 57i CT screens display the following during WAV streaming.
Streaming
00:10
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SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
To abort the WAV streaming, a user can press any of the following:
•Goodbye key
•Drop softkey
•Line key (not applicable to the 51i)
You can also place the handset (if in use) on hook.
RTP Recording and Simultaneous Playing (not supported on 51i)
The IP Phones now allow for RTP recording and simultaneous playing of an audio file via XML
Execute commands. An Administrator can use the RTP and multicast RTP commands (RTPTx,
RTPRx, RTPMTx, RTPMRx) with the “AastraIPPhoneExecute” object to initiate this feature.
The RTP and multicast RTP commands allow the phone to send/receive an RTP stream to/from
given multicast/unicast addresses (without involving SIP signaling). An Administrator can use the
following two options appended to the RTP commands for recording and simultaneously playing
of an audio file:
•mix - When appended to an RTP command, “mix” enables RTP recording and simultaneous
playing of an audio file.
•disableIcon - When appended to an RTP command, “disableIcon” disables the display of the
mixed call icons on the LCD of the phones.
Note: This feature is not supported on the 51i. Also, if the cordless
handset (CT) and the base phone are connected (Models 9480i CT and
57i CT), this feature is not supported.
Sending RTP
When a phone receives a request for sending RTP:
•and there is an existing call on the phone, no new audio session is created; the existing RTP
call (incoming or outgoing) is sent to the receiving party.
•and the phone is already in a 3-way conference, the request is declined.
•and there is no existing call on the phone, a new audio session is created and the RTP call is
sent to the receiving party.
The following scenarios assume that a request for sending mixed RTP is initiated.
Phone StateAction When RTP Transmitting
Phone is in idle state.The phone initiates a new RTP session on
the paging line and the paging line displays.
There is an existing voice call on the
phone.
The phone is in conference. The request for sending RTP is declined.
The phone is not in conference;
however, both voice and conference
streams are busy (cordless handset).
The phone starts sending the mixed RTP
stream, paging line does not display.
The request for sending RTP is declined.
RN-001029-02, Release 2.3, Rev 0031
IP Phone Release Notes 2.3
New Features in Release 2.3
Phone StateAction When RTP Transmitting
The phone is in a call using the
conference stream and the voice
stream is free.
The active voice call is dropped.
(Mixed RTP stream was being sent
using this voice call).
A new call comes in while an active
voice call is put on hold. (Mixed RTP
stream was being sent using this voice
call).
The phone is sending the mixed RTP
stream, and there is an existing voice
call on the phone. A request for
initiating localized conference is
initiated.
Note: When an RTP stream is being sent with RTP recording and
simultaneous playing (mix), the IP Phone LCD displays an icon with an
‘m’ on top to indicate that a mixed RTP stream is being sent.
Paging call is initiated using the voice
stream with the mixed audio from the
conference stream; paging line does not
display.
RTP stream is dropped.
RTP mixed stream is sent for the currently
active call.
Localized conference request is declined,
and the focus is given back to the initiating
line.
32RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
Receiving RTP
When a phone receives a request for receiving RTP:
•if there is an existing audio path, (handset, headset, or speakerphone), the phone uses it to play
the incoming RTP stream on top of the existing one.
•if the phone is already in a 3-way conference, the request is declined.
•if there is no existing audio path, the phone creates a new one (speakerphone) to play the
incoming RTP stream
The following scenarios assume that a request for receiving mixed RTP is initiated.
Phone StateAction When RTP Receiving
Phone is in idle state.The phone initiates a new RTP session on
the paging line and the paging line displays.
There is an existing voice call on the
phone.
The phone is in conference. The request for receiving RTP is declined.
The phone is not in conference;
however, both voice and conference
streams are busy (cordless handset).
The phone is in a call using the
conference stream and the voice
stream is free.
The active voice call is dropped. (RTP
stream was being played on top of this
voice call).
A new call comes in while an active
voice call is put on hold. (RTP stream
was being played on top of this voice
call).
The active voice call is dropped RTP stream is dropped.
The phone starts playing the incoming RTP
stream on top of the existing call. Paging
line does not display.
The request for receiving RTP is declined.
Incoming RTP stream is played on top of
the existing voice stream. Paging line does
not display.
RTP stream is dropped.
RTP mixed stream is played on top of the
currently active call.
Note: When RTP stream is being received with RTP recording and
simultaneous playing (mix), you can use the volume controls on the IP
phone to adjust the volume (increase or decrease) to your specifications.
RN-001029-02, Release 2.3, Rev 0033
IP Phone Release Notes 2.3
New Features in Release 2.3
Examples
The following are examples of using the “mix” and “disableIcon” execute commands with the
AastraIPPhoneExecute object. These commands apply to both the RTP and multicast RTP
XML commands (RTPTx, RTPRx, RTPMTx, RTPMRx).
•Send mix unicast RTP stream (if there is existing voice call) to 10.30.100.20:21000. Mix
icon is displayed on the phone’s call screen.
<AastraIPPhoneExecute>
<ExecuteItem URI="RTPTx:10.30.100.20:21000:mix">
</AastraIPPhoneExecute>
•Receive unicast RTP stream from 10.30.100.20 at port 21000 with the egress voice settings
at 3 levels more than the current offset and play on top of the existing voice stream (if any).
•Receive unicast RTP stream from 10.30.100.20 at port 21000 with the voice settings as used
earlier and play on top of the existing voice call (if any). Also, the mixing icon is not shown
on the call screen.
In previous releases, when an XML application was active on the phone, it suppressed all
dialpad events. This suppression was to prevent digit events from being sent to the underlying
line application.
On the IP Phones, an XML Developer can now control digit passthrough while the phone is in
the connected state. (This feature is only applicable to phones in the connected state). This can
be done by setting the new “allowDTMF” attribute with the XML objects
AastraIPPhoneTextScreen, AastraIPPhoneFormattedTextScreen, and
AastraIPPhoneImageScreen. Setting this attribute to “yes” allows dialpad events to pass
through the XML applications.
Note: The default behavior for this feature is to suppress dialpad events
when an XML object is in focus (same as in previous releases). Other
XML objects ignore this attribute.
34RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
Examples
•The following is an example of setting the allowDTMF attribute to allow dialpad events to
pass through the XML applications.
<AastraIPPhoneTextScreen allowDTMF="yes">
<Title>Test</Title>
<Text>Digits should pass through this object</Text>
</AastraIPPhoneTextScreen>
•The following is an example of setting the allowDTMF attribute to prevent dialpad events
from passing through the XML applications (this is the default setting).
<AastraIPPhoneTextScreen allowDTMF="no">
<Title>Test</Title>
<Text>Digits do not pass through this object</Text>
</AastraIPPhoneTextScreen>
Non-Blocking Action URI
The Action URI feature on the IP Phones is now enhanced to prevent the phone from hanging if
the Action URIs should fail. Also, support for transparent, non-blocking, XML post execute item
URI was added.
New XML URI Variables
Action URIs on the phone support variables, which are replaced with their corresponding value
when the URI is triggered. New XML variables have been added that can be used with an onhook
Action URI, and allow for enhanced information in call records and billing applications:
•$$LOCALIP$$ - Phone's IP address
•$$CALLDURATION$$ - Duration of the current/last call
•$$CALLDIRECTION$$ - Specifies whether the current/last call was incoming or outgoing
XML Web Applications Button
The IP Phones now have a feature that allows a User or Administrator to access XML-based
applications via a new pre-programmed softkey called, "WebApps".
On the 55i, 57i, 57i CT, 9480i, and 9480i CT phones, the WebApps softkey displays on softkey 1
on the idle screen.
On the 51i, 53i, and 9143i, you can access the WebApps option in the Services menu.
You can set any programmable or softkey as a WebApps key. Pressing this key redirects the phone
to a Web application provided by your Administrator.
RN-001029-02, Release 2.3, Rev 0035
IP Phone Release Notes 2.3
New Features in Release 2.3
XML Key Redirection
The IP phones now allow the redirecting of phone-based hard keys to XML scripts. This allows
the server to provide the phone with Redial, Transfer (Xfer), Conference (Conf), and Intercom
(Icom) key features, and the Voicemail option feature, rather then accessing them from the
phone-side. This feature allows you to access the redirected keys and voicemail option from the
server using the IP Phone’s Services Menu. By default, the server-side keys function the same as
the phone-side key features.
The following table identifies the phone states that apply to each key redirection.
Hard Keys/OptionsRedirects in
Conference (Conf)the connected state
Transfer (Xfer)the connected and dialing states
Redialall states
Intercom (Icom)all states
Voicemailall states
Notes:
1.If XML key redirection is enabled on the 51i., the Xfer and Conf
menu items perform redirection.
2.Key remapping takes precedence over redirecting.
3.Disabling the redial, conference, or transfer features on the phone
also disables the redirection of these keys.
The following URI configuration parameters control the redirection of the keys and the
voicemail option:
•redial script
•xfer script
•conf script
•icom script
•voicemail script
An Administrator can configure the XML key, redirection URI parameters using the
configuration files only.
36RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
Configuring XML Redirection of the Redial, Xfer, Conf, and Icom Keys, and the
Voicemail Option
Use the following script parameters to redirection the Redial, Xfer, Conf, and Icom keys, and the
Voicemail Option so that these features are provided by the server instead of the phone.
Parameter–
redial script
DescriptionSpecifies a redial script for the phone to use. When this parameter is set,
pressing the Icom key GETs the specified URI from the server to use in
performing the Intercom action.
Configuration Filesaastra.cfg, <mac>.cfg
set, selecting the voicemail option from the Services Menu GETs the
specified URI from the server instead of starting the Voicemail application.
38RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
Options Key Redirection
(for Options Menu on all phones and Services Menu on 51i)
The IP phones now allow the redirecting of the Options Key (Options Menu on all phones and
Services Menu on the 51i) to an XML script. This allows the server to provide the phone with
available options, rather then accessing them from the phone-side. You access the Options Key
XML script by pressing the Options Key. You can still access the Options Menu from the
phone-side by pressing and holding the Options key to display the phone-side Options Menu.
The following URI configuration parameter controls the redirection of the Options Key:
•options script
IMPORTANT NOTES
•If no Options URI script is configured, the local Options Menu on the phone displays as
normal.
•If you configure password access to the Options Menu, this password is required when
accessing the local Option Menu, but is not required for the Options Key redirection feature.
•Pressing the Options Menu for redirection from the server does not interfere with normal
operations of the phone (for example, pressing the options menu when on a call does not affect
the call).
•If the phone is locked, you must unlock the phone before accessing the Options Menu redirect
feature. After pressing the Options Key, the phone displays a screen that allows you to unlock
the phone before continuing.
•On the 51i, the redirection feature works after selecting “Options” from the Services Menu. To
display the original Options Menu, press and hold the Services key.
An Administrator can configure the XML Options Key (or Services Key on the 51i), redirection
URI parameter using the configuration files only.
RN-001029-02, Release 2.3, Rev 0039
IP Phone Release Notes 2.3
New Features in Release 2.3
Configuring XML Redirection of the Options Key (Services Key on the 51i)
Use the following parameter to configure XML redirection of the Options Key (or Services Key
on the 51i).
Parameter–
options script
DescriptionSpecifies an Options script for the phone to use. When this parameter is set,
pressing the Options Key (or Services Key on the 51i) GETs the specified
URI from the server.
Note: Pressing and holding the Options key (or the Services Key on the 51i)
displays the local Options Menu on the phone.
40RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
XML Applications and Off-Hook Interaction
In Release 2.3, a feature has been implemented that prevents the phone from going into the
off-hook/dialing state when the handset is off-hook and the call ends.
In previous releases, the phone behaved as in the following scenario:
You are in a call using the handset and the phone displays an XML application. The far-end
terminates the call, and a new XML application gets pushed/pulled onto the display. Since the
handset is off-hook and in idle mode, the "offhook idle timer" starts. When this timer expires, the
phone applies dial tone and moves to the off-hook/dialing state, which then destroys the XML
application that was being displayed.
In Release 2.3, you can set an “auto offhook” parameter that determines whether or not the phone
is prevented from entering the off-hook/dialing state, if the handset is off-hook and the call ends.
An Administrator can enable or disable the “auto offhook” parameter using the configuration files
only.
Configuring the Off-Hook Interaction Feature
Use the following parameter to configure the XML application and off-hook interaction feature.
Parameter–
auto offhook
DescriptionSpecifies whether or not the phone is prevented from entering the off-hook/
FormatBoolean
Default Value0 (disabled)
Range0 (disabled)
Exampleauto offhook: 1
Configuration Filesaastra.cfg, <mac>.cfg
dialing state, if the handset is off-hook for more than 2 seconds, and the call
ends.
1 (enabled)
RN-001029-02, Release 2.3, Rev 0041
IP Phone Release Notes 2.3
New Features in Release 2.3
XML URI for Key Press Simulation
Release 2.3 provides a feature that allows an XML Developer or Administrator to define XML
Key URIs that can send key press events to the phone, just as if the physical hard key, softkey, or
programmable key were pressed on the phone.
When the Key URI event is sent from the server to the phone, the phone initiates the event as if
the key was physically pressed. If the key is not present on the phone (hard key) or not available
(softkey or programmable key), when the phone receives the URI, the event is discarded. If you
are in the process of changing the softkey or programmable key setting, or the key is disabled
while the event is being processed, the request is discarded. The phone maps key events to it’s
physical keys and not to it’s mapped logical keys.
The following table identifies the XML URIs for pressing buttons on the phone..
XML Key URIDescription
Line Keys
Key:Line1 to Key:Line4 Line 1 to 4 Keys
Keypad Keys
Key:KeyPad0 to Key:KeyPad9 Numeric Keypad Keys 0-9
Key:KeyPadStar * - Star Key
Key:KeyPadPound # Hash Key
Softkeys
Key:SoftKey1 to Key:SoftKey<n> Softkey 1 to <n> (valid softkeys depend on
Key:TopSoftKey1 to Key:TopSoftKey<n>
top
Programmable Keys
Key:PrgKey1 to Key:PrgKey<n> Programmable keys 1 to <n> (valid
Expansion Module Keys
Key:ExpMod1SoftKey1 to
Key:ExpMod1SoftKey60
Key:ExpMod2SoftKey1 to
Key:ExpMod2SoftKey60
Key:ExpMod3SoftKey1 to
Key:ExpMod3SoftKey60
Note: The phone ignores URI line keys 5 to
9 since it does not have Line 5 to 9 physical
keys.
the number of physical softkeys on the
phone)
Top softkeys 1 to <n> ((valid top softkeys
depend on the number of physical top
softkeys on the phone)
programmable keys depend on the number
of physical programmable keys on the
phone)
Expansion module 1 softkeys 1 to 60
Note: The phone ignores URI expansion
module key events if the keys are not
physically present on the expansion
module.
Expansion module 2 soft keys 1 to 60
Expansion module 3 soft keys 1 to 60
42RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
XML Key URIDescription
Vol u m e K e y
Key:VolDwn Volume Decrease Key
Key:VolUp Volume Increase Key
Feature Keys
Key:Xfer Transfer Key
Key:Conf Conference Key
Key:Services Services Key
Key:Intercom Intercom Key
Key:Headset Headset Key
Note: For Headset URI key, the behavior
will be as if the "speaker/headset" key is
pressed; and does not switch to headset for
headset key event or to speaker for speaker
key event.
Key:Speaker Speaker Key
Note: For Speaker URI key, the behavior
will be as if the "speaker/headset" key is
pressed; and does not switch to headset for
headset key event or to speaker for speaker
key event.
Key:MuteMute Key
Key:Hold Hold Key
Key:Redial Redial Key
Key:Callers Callers Key
Key:Directory Directory Key
Key:Options Options Key
Key:Save Save Key
Key:Delete Delete Key
Key:Swap Swap Key
Key:Goodbye GoodBye Key
Navigation Keys
Key:NavUp Navigation Up Key
Key:NavDwn Navigation Down Key
Key:NavLeft Navigation Left Key
Key:NavRight Navigation Right Key
Function Keys (only if physically configured on the phone or expansion module)
KeyParkPark Softkey
KeyPickupPickup Softkey
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IP Phone Release Notes 2.3
New Features in Release 2.3
Notes:
1.If the URI key is a valid key, the phone executes the key regardless
of the current state on the phone.
2.Park and Pickup XML URI softkeys are available ONLY if these
features are physically configured on the phone or expansion module.
Examples
There are two ways to format the XML key URI:
For XML Post Messages
<ExecuteItem URI="<XML Key URI>" />
Example:
<ExecuteItem URI="Key: Line1" />
For XML Key Scripts
<URI><XML Key URI></URI>
Example:
<URI>Key: Line1</URI>
<SoftKey index="1">
<Label>Keypad1</Label>
<URI>Key: Line1</URI>
</SoftKey>
44RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
Using XML Commands to Reset Local Data on the Phone
Release 2.3 provides new XML commands that allow an XML Developer to delete and reset the
phone’s directory, callers list, redial list, and the local.cfg file. You use these commands with the
AastraIPPhoneExecute object. The following table identifies the new XML commands to reset
data on the phone.
XML CommandDescription
Command: ClearLocal•Deletes the local.cfg file
•Displays a "Resetting" message to the
IP Phone UI
•Resets the phone
Command: ClearCallersList •Deletes the Callers List
•Clears any missed call messages
Command: ClearDirectory •Deletes the Directory
Command: ClearRedialList•Deletes the Redial List
Examples
The following examples illustrate the use of the new XML commands to delete and reset data on
the phone. The command is shown in bold.
<AastraIPPhoneExecute>
<ExecuteItem URI="Command: ClearLocal"/>
</AastraIPPhoneExecute>
<AastraIPPhoneExecute>
<ExecuteItem URI="Command: ClearCallersList"/>
</AastraIPPhoneExecute>
<AastraIPPhoneExecute>
<ExecuteItem URI="Command: ClearDirectory"/>
</AastraIPPhoneExecute>
<AastraIPPhoneExecute>
<ExecuteItem URI="Command: ClearRedialList"/>
</AastraIPPhoneExecute>
Note: The XML commands to reset local data on the phone are not
applicable to the Model CT handsets.
RN-001029-02, Release 2.3, Rev 0045
IP Phone Release Notes 2.3
New Features in Release 2.3
Action URI Disconnected Feature
A new parameter, “action uri disconnected” has been added to Release 2.3 that allows a
disconnect event to occur when the phone transitions from any active state (outgoing, incoming,
connected, or calling) to an idle state. A new line variable called “$$LINESTATE$$” has also
been added to Release 2.3 that an Administrator can use with the “action uri disconnected”
parameter as well as with existing XML URI parameters.
Note: The $$LINESTATE$$ variable is optional and not required when
enabling the “action uri disconnected” parameter.
If the Administrator enables this feature (by specifying a disconnect URI), when a call is
disconnected, the phone
phones finds a configured URI with a $$LINESTATE$$ variable, it replaces the
$$LINESTATE$$ variable with the appropriate line state of the current active line. After all of
the variables are bound, the phone executes a GET on the URI. The following table lists the
applicable values for the $$LINESTATE$$ variable.
$$LINESTATE$$ Value DescriptionMeaning in a Disconnected URI
checks to see if the event has a Disconnect URI configured. If the
IDLEPhone is idle.N/A
DIALINGPhone is offhook and ready to dial.N/A
CALLINGA SIP INVITE was sent but no
response was received.
OUTGOINGRemote party is ringing.Call was cancelled.
INCOMINGLocal phone is ringing.Call was missed or cancelled.
CONNECTEDParties are talking.Call was successful.
CLEARINGCall was released but not
acknowledged.
Error occurred during the call.
N/A
The Action URI Disconnect feature allows an Administrator to determine the reason for the
disconnect if required.
Note: If you enable the Action URI Disconnect feature by specifying a
URI, the URI is called when any disconnect event occurs including an
intercom call or a conference setup.
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SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
Example
If you enter the following string on Phone A for the “action uri disconnected” parameter:
action uri disconnected: http://fargo.ana.aastra.com/
disconnected.xml?state=$$LINESTATE$$
and then Phone A calls Phone B, Phone B answers and then hangs up, Phone A executes a GET
on:
An Administrator can enable the disconnect feature using the configuration files or the Aastra Web
UI.
Configuring the Action URI Disconnect Feature Using the Configuration Files
Use the following parameter to configure the Action URI Disconnect feature.
Parameter–
action uri disconnected
DescriptionSpecifies the URI that the phone executes a GET on, when it transitions from
FormatString
Default ValueBlank
RangeAny valid URI
Exampleaction uri disconnected: http://fargo.ana.aastra.com/
Aastra Web UIAdvanced Settings->Action URI->Event
Configuration Filesaastra.cfg, <mac>.cfg
the incoming, outgoing, calling, or connected state into the idle state.
This parameter uses the following variable to determine the state of the line:
$$LINESTATE$$
disconnected.xml?state=$$LINESTATE$$
RN-001029-02, Release 2.3, Rev 0047
IP Phone Release Notes 2.3
New Features in Release 2.3
Configuring the Action URI Disconnect Feature Using the Aastra Web UI
Use the following procedure to configure the Action URI Disconnect feature.
Aastra Web UI
1Click on Advanced Settings->Action URI->Event.
2In the “Disconnected” field, enter a valid URI for which the phone executes a GET on, when it transitions from
the incoming, outgoing, calling, or connected state into the idle state. Leaving this field empty disables the Action
URI Disconnected feature. For example,
The following table lists the applicable values and descriptions for the $$LINESTATE$$.
$$LINESTATE$$ ValueDescriptionMeaning in a Disconnected URI
IDLEPhone is idle.N/A
DIALINGPhone is offhook and ready to
dial.
CALLINGA SIP INVITE was sent but no
response was received.
OUTGOINGRemote party is ringing.Call was cancelled.
INCOMINGLocal phone is ringing.Call was missed or cancelled.
CONNECTEDParties are talking.Call was successful.
CLEARINGCall was released but not
acknowledged.
3Click to save your settings.
Error occurred during the call.
N/A
N/A
48RN-001029-02, Release 2.3, Rev 00
User Interface Features
Preferred Line Focus Feature
In previous releases of the IP Phones, after a call (incoming or outgoing), the phone would stay on
the line that was used for that call. For example, if you made a call on Line 2, then after the call,
the display would be showing line 2. Then when you picked up the handset, the phone would go
offhook on line 2.
In Release 2.3, an Administrator or User can now define a preferred line as well as a preferred timeout. If a preferred line is selected, after a call ends (incoming or outgoing), the display
switches back to the preferred line. Next time you go off-hook, you pickup on the preferred line.
You can specify the number of seconds it takes for the phone to switch back to the preferred line
using the “preferred timeout” parameter.
An Administrator can configure the “preferred line” and the “preferred timeout” parameters
using the configuration files or the Aastra Web UI. A User can configure these parameters using
the Aastra Web UI only.
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
The following table provides the behavior of the preferred line focus feature with other features on
the phone.
Phone FeaturePreferred Line Focus Behavior
call returnThe phone switches back to the focused line immediately after
the call ends.
speeddialThe line is already specified when the speeddial is created. The
phone switches back immediately after the call ends.
conferenceFor incoming calls, the phone switches back immediately after
the call ends.
transferFor incoming or outgoing calls, the current behavior is that the
same line used to transfer the call does not change. For incoming
calls, the phone switches back immediately after the call
transfers.
blfThe phone switches back immediately after the call ends.
parkThe phone switches back immediately after the call ends.
voicemailThe phone switches back immediately after the call ends.
redialThe phone switches back immediately after the call ends.
dialingFor incomplete dialing on a non-preferred line, the focus does not
change if some digits are entered.
If no digits are entered or digits were cleared, the focus changes
to preferred line after the time out has passed without activities.
RN-001029-02, Release 2.3, Rev 0049
IP Phone Release Notes 2.3
New Features in Release 2.3
Phone FeaturePreferred Line Focus Behavior
caller idIf the "Switch UI Focus To Ringing Line" parameter is disabled,
factory defaultFactory default and recovery mode clears the "preferred line" and
Notes:
1.If you specify a value of “0” for the preferred line parameter, it
disables the preferred line focus feature. The phone behaves as in
previous releases.
2.If you specify a value of “0” for the preferred line timeout
parameter, the phone returns the line to the preferred line immediately.
Configuring Preferred Line Focus via the Configuration Files
the User is able to see the Caller ID when the phone switches the
focus to the ringing line.
"preferred line timeout" parameters, and the phone operates in a
non-preferred line mode.
Use the following parameters to configure preferred line focus using the configuration files.
Parameter–
preferred line
Preferred Line
(in Web UI)
DescriptionSpecifies the preferred line to switch focus to when incoming or outgoing
FormatInteger
Default Value1
Range0 (none - disables the preferred line focus feature)
Examplepreferred line: 2
Parameter–
preferred line timeout
Preferred Line Timeout
(seconds)
(in Web UI)
DescriptionSpecifies the time, in seconds, that the phone switches back to the preferred
FormatInteger
Default Value0 (the phone returns the line to the preferred line immediately)
Range0 to 999
Examplepreferred line timeout: 30
Configuration Filesaastra.cfg, <mac>.cfg
Aastra Web UIBasic Settings->Preferences->General
calls end on the phone.
1 to 9
Configuration Filesaastra.cfg, <mac>.cfg
Aastra Web UIBasic Settings->Preferences->General
line after a call (incoming or outgoing) ends on the phone, or after a duration
of inactivity on an active line.
50RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
Configuring Preferred Line Focus via the Aastra Web UI
Use the following parameters to configure preferred line focus using the Aastra Web UI.
Aastra Web UI
1Click on Basic Settings->Preferences->General.
New Features in Release 2.3
Preferred Line
Preferred Line
Timeout
2In the “Preferred Line” field, select a preferred line to switch focus to after incoming or outgoing calls end on the
phone. Valid values are:
•None (disables the preferred line focus feature)
•1 to 9
Default is 1.
For example, if you set the preferred line to “1”, when a call (incoming or outgoing) ends on the phone (on any line),
the phone switches focus back to Line 1.
3In the “Preferred Line Timeout” field, enter the amount of time, in seconds, that the phone switches back to the
preferred line after a call (incoming or outgoing) ends on the phone, or after a duration of inactivity on an active
line. Default is 0. Valid values are:
•0 to 999
4Click to save your changes.
RN-001029-02, Release 2.3, Rev 0051
IP Phone Release Notes 2.3
New Features in Release 2.3
Dialpad Speeddial Supported on All Phones
Previously, only the 51i IP Phone supported a dialpad speeddial feature using the IP Phone UI or
the Aastra Web UI. In Release 2.3, all the phones now support this feature. Using the IP Phone
UI or the Aastra Web UI, an Administrator or User can create speeddial keys on the dialpad.
For more information about speeddial keys, see your model-specific User Guide.
Creating a Speeddial Key using the IP Phone UI
In addition to creating speeddial keys by pressing and holding a key on the keypad, you can also
select “Speed Dial Edit” at the path Options->Preferences->Speed Dial Edit. Use the following
procedure to create speeddial keys using the IP Phone UI
You can cancel out of the speeddial editing process at any time without saving, by pressing the
key.
Aastra IP Phone UI
StepAction
1Press Options, and then select Preferences.
On the 51i:
Press Services, and then select Options->Preferences.
2Select Speed Dial Edit.
For the 51i (1 line only)
3The following prompt displays:
“Press a SD (speeddial) button”.
4Press any number key on the keypad. For example, “5”.
Note: If a number on the keypad is already setup as a speeddial key, pressing the applicable number when
creating a speeddial key displays the speeddial information for you to edit.
The following prompt displays:
“Enter Number”
5Enter the number to assign to this speeddial key and press SAVE.
Note: On the 51i, you can remove the Speeddial Key by erasing the speeddial number digits (leaving the
speeddial value blank) and then press SAVE or the RIGHT arrow key. Use the LEFT arrow key to delete the digits.
The following prompt displays:
“Saved Memory Key <phone number>”.
52RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
Aastra IP Phone UI
StepAction
For the 53i and 9143i
3The following prompt displays:
“Press a SD (speeddial) button”.
4Press any number key on the keypad. For example, “5”.
Note: If a number on the keypad is already setup as a speeddial key, pressing the applicable number when
creating a speeddial key displays the speeddial information for you to edit.
The following prompt displays:
“Enter Number”
5Enter the number to assign to this speeddial key and press SAVE.
6Select the Line number for which this speeddial dials out on., and press SAVE.
The following prompt displays:
“Saved Memory Key <phone number>”.
New Features in Release 2.3
For the 55i, 57i, 57i CT, 9480i, and 9480i CT
3The following prompt displays:
“Press a SD (speeddial) button”.
4Press any number key on the keypad. For example, “5”.
Note: If a number on the keypad is already setup as a speeddial key, pressing the applicable number when
creating a speeddial key displays the speeddial screen for you to edit.
A screen displays allowing you to enter the Name, Number, and Line for which to assign the speeddial key.
5In the “Enter Name” field, enter a name to assign to the speeddial key and press the DOWN arrow.
6In the “Enter Number” field, enter a phone number to assign to the speeddial key and press the DOWN arrow.
7In the “Line” field, use the UP and DOWN arrow keys to select a specific line for which the speeddial dials out on.
8Press SAVE to save the changes for the new speeddial key.
RN-001029-02, Release 2.3, Rev 0053
IP Phone Release Notes 2.3
New Features in Release 2.3
Creating a Speeddial Key using the Aastra Web UI
Use the following procedure to create a speeddial key using the Aastra Web UI.
Aastra Web UI
1Click on Operation->Keypad Speed Dial.
2Choose a key for which to assign a speeddial key. In the “Value” field, enter the phone number to assign to the
speeddial key.
3Click to save your changes.
54RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
UTF- 8 Codec for Multi-National Language Support
The IP Phones, expansion modules, and cordless handsets previously supported ISO 8859-1
(Latin1) language. The IP Phones and expansion modules now include support for ISO 8859-2
(Latin2) of multi-national languages when displaying and inputing in the IP Phone UI and the
Aastra Web UI.
Note: This feature is not applicable to the handsets on the 57i CT and the
9480i CT.
UTF-8 is also compatible with XML encoding on the IP Phones.
The following table illustrates the Latin 2 character set now used on the IP Phones.
RN-001029-02, Release 2.3, Rev 0055
IP Phone Release Notes 2.3
New Features in Release 2.3
Addition of New Timezone and Country Codes
In Release 2.3, the following new timezones and country codes have been added.
Time Zone Country Code/Time
Zone Name
AZ - Azerbaijan (Baku)AZT
GE - Georgia (Tbilisi)GET
MU - MauritiusMUT
OM - Oman (Muscat)GST
AE - United Arab Emirates (Dubai)GST
Time Zone Code
56RN-001029-02, Release 2.3, Rev 00
SIP Features
P-Asserted-Identity (PAI) Support in UPDATE message
In previous releases, the IP Phones supported P-Asserted-Identity (PAI). In Release 2.3, the
phones now support PAI header in the UPDATE message, according to
draft-ietf-sipping-update-pai-00. This feature is always enabled.
If an UPDATE is received with a PAI header from a trusted source, the phone updates the display
with this information. The phone ignores any PAI received from untrusted entities.
DTMF Tones in Info Requests
In Release 2.3, the phones now support decoding and playing out DTMF tones sent in SIP INFO
requests. The following is now supported:
•Support signals 0-9, #, *
•Support durations up to 5 seconds
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
RN-001029-02, Release 2.3, Rev 0057
IP Phone Release Notes 2.3
New Features in Release 2.3
Ignore Out of Sequence Errors
In Release 2.3, an Administrator can configure the phone via the “sip accept out of order
requests” parameter to ignore CSeq number errors on all SIP dialogs on the phone. When this
parameter is enabled, the phone no longer verifies that the sequence numbers increase for each
message within a dialog, and does not report a "CSeq Out of Order" error if they do not increase.
Note: As the default Asterisk configuration does not fully track dialogs
through a reboot, it is recommended that this parameter be enabled when
using the BLF feature with an Asterisk server. If you do not enable this
feature, then rebooting the Asterisk server may cause BLF to stop
working. With this parameter enabled, the BLF key starts working again
when the phone re-subscribes, which by default, are one hour apart.
Enabling/Disabling “Out of Order SIP Requests”
Use the following parameter to enable or disable the SIP out of order requests.
Parameter–
sip accept out of order
requests
DescriptionEnables a workaround for non-compliant SIP devices (for example, Asterisk)
FormatBoolean
Default Value0 (disabled)
Range0 (disabled)
Examplesip accept out of order requests: 1
Configuration Filesaastra.cfg, <mac>.cfg
which do not increment the CSeq numbers in SIP requests sent to the
phone.
1 (enabled)
BL
58RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
SIP BLA Expires Timer
Release 2.3 now includes a SIP BLA subscription period parameter that allows an Administrator
to set the amount of time, in seconds, of the BLA subscription period.
If this parameter is set to zero (0), the phone uses the value specified for the BLA expiration in the
subscribe message received from the server. If no value is specified in the Subscribe message
received from the server, the phone uses the default value of 300 seconds.
You can configure this parameter using the configuration files or the Aastra Web UI.
Configuring SIP BLA Expirey Timer Using the Configuration Files
Use the following parameters to configure the BLA expirey timer using the configuration files.
Parameter–
sip bla subscription
period
BLA Subscription Period
(in Web UI)
DescriptionSpecifies the amount of time, in seconds, that the phone waits to receive a
FormatInteger
Default Value300
Range0 to 3700
Examplesip bla subscription period: 0
Configuration Filesaastra.cfg, <mac>.cfg
Aastra Web UIAdvanced Settings->Global SIP->
Advanced SIP Settings
BLA subscribe message from the server. If you specify zero (0), the phone
uses the value specified for the BLA expiration in the subscribe message
received from the server. If no value is specified, the phone uses the default
value of 300 seconds.
Note: When set to zero (0), the phone uses BLA expiry value specified in
subscribe message.
RN-001029-02, Release 2.3, Rev 0059
IP Phone Release Notes 2.3
New Features in Release 2.3
Configuring SIP BLA Expirey Timer Using the Aastra Web UI
Use the following procedure to configure the SIP BLA Subscription Period using the Aastra
Web U I .
Aastra Web UI
1Click on Advanced Settings->Global SIP->Advanced SIP Settings.
2In the “BLA Subscription Period” field, enter a value, in seconds, that the phone waits to receive a BLA subscribe
message from the server. If you specify zero (0), the phone uses the value specified for the BLA expiration in the
subscribe message received from the server. If no value is specified, the phone uses the default value of 300
seconds. Valid values are 0 to 3700. Default is 300 seconds.
3Click to save your changes.
60RN-001029-02, Release 2.3, Rev 00
Troubleshooting Features
WatchDog Task Feature
Release 2.3 provides a troubleshooting feature called the “WatchDog” task that monitors the
status of the phones and provides the ability to get stack traces from the last time the phone failed.
When the phone detects a failure (i.e., a crash), it automatically reboots. You can view a WatchDog
crash file using the Aastra Web UI at the path, Advanced Settings->Troubleshooting. You can
enable/disable the WatchDog task using the configuration files or the Aastra Web UI.
Enabling/Disable WatchDog Using the Configuration Files
Use the following parameter to enable/disable the WatchDog task for the IP Phones using the
configuration files.
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
New Features in Release 2.3
Parameter–
watchdog enable
WatchDog
(in Web UI)
DescriptionEnables/disables the use of the WatchDog task for the IP Phones.
FormatBoolean
Default Value1 (enabled)
Range0 (disabled)
Examplewatchdog enable: 0
Configuration Filesaastra.cfg, <mac>.cfg
Aastra Web UIAdvanced Settings->Troubleshooting
1 (enabled)
RN-001029-02, Release 2.3, Rev 0061
IP Phone Release Notes 2.3
New Features in Release 2.3
Enabling/Disable WatchDog Using the Aastra Web UI
Use the following procedure to enable/disable the WatchDog task for the IP Phones using the
Aastra Web UI. You can also view the “Crash Log” generated by the WatchDog task using the
Aastra Web UI.
Aastra Web UI
1Click on Advanced Settings->Troubleshooting.
Enable/Disable WatchDog
Enable/Disable WatchDog Task
2The “WatchDog” field is enabled by default. To disable the WatchDog task, click in the “Enabled” box to clear the
check mark.
3Click to save your changes.
View the Crash Log
4To view a crash log, in the “Get a Crash Log” field, click the SAVE AS button. You can open the file immediately, or
you can save the Crash Log to your PC.
62RN-001029-02, Release 2.3, Rev 00
View Crash Log
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
Issues Resolved in Release 2.3
Issues Resolved in Release 2.3
This section describes the issues resolved on the 9143i, 9480i, 9480i CT, and the 5i Series IP
phones in Release 2.3. The following table provides the issue number and a brief description of
each fix.
Note: Unless specifically indicated, these resolved issues apply to all
phone models.
Issue NumberDescription of Fix
User Interface
CLN1037853i: Pickup key now works when there is a held call on the phone.
CLN10430The TRUN password is no longer displayed in clear text on the WebUI.
ENH08045Phone now plays DTMF tones it receives via SIP INFO packets.
DEF09452Lock item is only displayed if both TLS and SRTP are in use.
DEF09517Dial plan is no longer used after receipt of a 484 Address Incomplete message.
DEF09689Audible ringback now continues to play if a monitored BLF line changes or an incoming call is
received on the phone.
DEF09879Corrected some German translation errors for the softkeys.
DEF09974When you are offhook and another call comes in, then the BLF/Xfer and Speeddial/Xfer keys work as
speed dials.
DEF10114Dialed numbers beginning with a * and having a total of three * (eg *8*81*201) are no longer treated
like an IP address.
DEF10125Early media is now played when STUN is enabled.
DEF1023651i: Dialing numbersfrom the directory that were entered into the directory from the phone now work
correctly.
DEF10369A transferred call (INVITE with replaces) no longer un-mutes the microphone.
XML
DEF07935When scrolling quickly through multiple input fields, the phone no longer switches back to the
previous screen.
DEF08582Pressing the right-arrow on a field that is not editable no longer exits the object.
ENH10109HTTPS URLs are now supported in the XML execute object.
SIP
DEF07398SCA: Phone now sends the correct appearance index for conference calls.
DEF08024BLF: Phone now retries SUBSCRIBE messages more often.
DEF08179ACD: Incorrect state change messages no longer trigger the phone to change state..
DEF08594Phone now rejects incoming SRTP calls if SRTP is disabled.
DEF08643BLA: Phone now sends a terminating NOTIFY when it terminates the subscription
DEF09820As-feature-sync SUBSCRIBE messages now accept the symmetric sip messaging parameter.
DEF10067Phone now supports usernames with dots (.)
RN-001029-02, Release 2.3, Rev 0063
IP Phone Release Notes 2.3
Issues Resolved in Release 2.3
Issue NumberDescription of Fix
DEF10102Retry-After headers with additional parameters are now supported.
DEF10323RTCP packets are no longer treated like early media.
DEF10328Phone now supports Replaces headers from servers that get the from-tag and to-tag mixed up.
Robustness
DEF07375Fixed a crash if you dial a number ending with a # using manual dialling.
DEF09481Phone now continues to attempt to get DHCP address in the background if it boots up prior to the
network being up.
DEF10185Phone no longer reloads firmware every time it is factory defaulted
64RN-001029-02, Release 2.3, Rev 00
SIP IP Phone Models 9143i, 9480i, 9480i CT and 5i Series Phones Release Note 2.3
Contacting Aastra Telecom Support
If you’ve read this release note, and consulted the Troubleshooting section of your phone model’s
manual and still have problems, please send inquiries via email to beta_support@aastra.com.
Contacting Aastra Telecom Support
RN-001029-02, Release 2.3, Rev 0065
IP Phone Release Notes 2.3
Contacting Aastra Telecom Support
66RN-001029-02, Release 2.3, Rev 00
Generic SIP IP Phone
Models 9143i, 9480i, 9480i CT, and 5i Series