Aastra Telecom IP Phone 480I User Manual

480i, 480i CT, 9112i, 9133i  
SIP IP PHONE  
41-001024-00  
Rev 09  
Release Notes  
Release 1.4.1  
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Contents  
XML: HTTP Refresh Header ..............................................................................33  
Backup Proxy/Registrar Support ...............................................................................34  
How it Works .......................................................................................................34  
Configuring via the Configuration Files ...............................................................34  
Auto-discovery Using mDNS .....................................................................................39  
41-001024-00, Rev 09, Release 1.4.1  
iii  
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Configuring via the Configuration Files ...............................................................46  
Description ................................................................................................................50  
Enhancements/Changes ...........................................................................................51  
Known Anomalies in 1.4.1 ............................................................................................52  
Description ................................................................................................................52  
Contacting Aastra Telecom Support............................................................................ 53  
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SIP IP Phone  
Models 480i, 480i CT, 9112i, 9133i  
Release Notes 1.4.1  
This document provides information specific to the SIP IP Phone release 1.4.1. It  
General Information (release content, hardware supported, bootloader  
New Features in 1.4.1  
Issues Resolved in Release 1.4.1  
Known Anomalies in 1.4.1  
Contacting Aastra Telecom Support  
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IP Phone Release Notes 1.4.1  
General Information  
General Information  
Release Content Information  
This document provides release content information on the Aastra 480i, 480i CT,  
9112i, and 9133i SIP IP phone firmware.  
Model  
480i  
Release Name  
Release Version  
Release Filename  
Release Date  
Generic SIP  
Generic SIP  
Generic SIP  
Generic SIP  
1.4.1  
1.4.1  
1.4.1  
1.4.1  
FC-000032-01-09  
FC-000040-00-09  
FC-000058-01-09  
FC-000046-01-09  
November 2006  
November 2006  
November 2006  
November 2006  
480i CT  
9112i  
9133i  
Hardware Supported  
This release of firmware is compatible with the following Aastra IP portfolio  
products:  
480i  
480i CT  
9112i  
9133i  
Bootloader Requirements  
This release of firmware is compatible with the following Aastra IP portfolio  
product bootloader versions:  
480i - Bootloader 1.1.0.4 or above  
480i CT - Bootloader 1.1.0.4 or above  
9112i - Bootloader 1.1.0.10 or above  
9133i - Bootloader 1.1.0.10 or above  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
General Information  
Upgrade Notes  
This section provides notes that customers should be aware of before upgrading to  
IP phone firmware release 1.4.1. Aastra Telecom recommends that customers read  
this section completely and thoroughly prior to the upgrade to avoid any known  
issues with the upgrade process.  
The recommended base IP Phone firmware to upgrade to this new firmware  
release is 1.2.2 or above.  
Users who previously adjusted the audio transmit and receive gains for the  
Aastra IP Phones should reset the values to "0" and re-tune their phones again  
with release 1.4.1 if necessary. This is required to avoid high transmit levels  
heard by the far-end after upgrading to 1.4.1. To set audio transmit and  
receive gains in the 1.4.1 configuration files, see the SIP IP Phone  
Administrator Guide Release 1.4 or the SIP IP Phone Release Notes Version  
1.3.1.  
Users upgrading from release 1.2.2 to 1.4.1 may experience post-upgrade  
configuration issues IF they previously configured their phones solely using  
the telephone user interface or Aastra Web user interface. Users may need to  
reconfigure their settings again after the upgrade due to new file system  
enhancements. This issue does not affect customers who configure their  
phones using a TFTP/FTP/HTTP server.  
Users who wish to downgrade from release 1.4.1 to 1.2 must downgrade to  
release 1.2.2 prior to downgrading to release 1.2, in order to restore file  
system structure. Failure to downgrade to version 1.2.2 first may result in  
configuration parameters being ignored.  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
New Features in 1.4.1  
Description  
This section describes the features new to the IP phones in release 1.4.1. The  
features apply to all IP Phone models (480i, 480i CT, 9112i, and 9133i), unless  
specifically stated that only a particular IP phone model supports the feature.  
The following table lists the new features in release 1.4.1. The paragraphs  
following this table describe each feature in detail.  
Feature  
Description  
Page  
All Models  
SIP Timers  
The following are SIP timers that were added or changed  
on the IP phones:  
SIP Registration Renewal Timer - A new parameter called sip page 7  
registration renewal timer has been added that allows an  
administrator to control when the phone renews SIP  
registrations. Configurable via configuration files and Aastra  
Web UI.  
now set the time, in seconds, that the phone waits until it  
attempts to register the phone after a REGISTER message  
times outs. This new parameter is called sip registration  
timeout retry timer. Configurable via configuration files and  
“SIP Registration Retry Timer” Web UI Parameter Name  
Change - In the Aastra Web UI only, the “registration retry  
timer” parameter name has changed to registration failed  
retry timer.  
page 10  
Explicit Message Waiting Indicator (MWI) Subscription Period page 11  
- An administration can now set the length of time, in  
seconds, of an explicit MWI subscription. This value allows  
the phone to re-subscribe the explicit MWI subscription  
before the timeout value is reached.  
Prefix Dialing  
An administrator can now enter a digit(s) at the end of the  
Local Dial Plan parameter string, which allows the IP phone  
to automatically add the digit as a prefix to a dialed number.  
page 13  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Feature  
Description  
Page  
Last Number Redial  
A user can now press the REDIAL key to dial the last number page 15  
dialed. They can also press the REDIAL button once, scroll  
the list of numbers, then press the REDIAL button again to  
dial the number that displays on the screen.  
XML Objects and  
Enhancements  
The following are new XML objects on the IP phone:  
XML: AastraIPPhoneStatus object - The IP phones now  
display a status message on a single designated line on  
the phone’s idle screen when XML information is pushed  
from the servers.  
page 16  
The 480i/480i CT phones display messages on the  
second line (where “No Service” would display). The  
9112i/9133i phones display messages on the first line  
(overriding the DisplayName). Long messages that are  
wider then the phone screen get truncated.  
XML: AastraIPPhoneExecute Object - This object  
provides the ability to execute commands on the phone  
using XML. Specific commands you can use with this  
object are Reset and NoOp.  
page 23  
page 25  
The following are XML enhancements on the IP phone:  
XML: Action URI - This feature provides administrators  
the ability to specify a URI for the phone to GET when  
certain events occur.  
XML: Softkey URI - This feature allows the user to specify page 30  
variables in the XML softkey URIs that are bound when  
the key is pressed.  
XML: HTTP Refresh Header - All current XML screen  
objects now have the ability to be refreshed by adding a  
Refresh and URL setting to the HTTP headers.  
page 33  
Backup Proxy/Registrar  
Support  
The IP phones now support the use of a backup SIP proxy/  
registrar. You can configure this feature on a global or per-line  
basis via the configuration files or the Aastra Web UI.  
page 34  
Auto-discovery Using  
mDNS  
Release 1.4.1 introduces a process that allows the phones to page 39  
auto-discover a TFTP server using mDNS, and subsequently  
be automatically configured by a TFTP server.  
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IP Phone Release Notes 1.4.1  
Feature  
Description  
Page  
IP Phone Features for  
Sylantro Servers  
The following are new features added to the IP phones  
for Sylantro servers:  
Last Call Return (lcr) Support - A new feature has been  
added to the IP phones that allow a user or administrator to  
configure a "last call return" function on a softkey or  
programmable key. This feature is for Sylantro servers only.  
You can configure the “lcr” softkey feature via the  
configuration files and the Aastra Web UI.  
page 40  
Support for additional “Alert Info” keywords for distinctive  
ringing - The IP phones now allow you to configure new  
distinctive ringing priority alert parameters for Sylantro  
servers. These “info” parameters allow you to configure  
specific priority alert tones for each parameter that may  
appear as key words in the “Alert-Info” header of a Sylantro  
server. The new keywords are alert-acd, alert-community-1,  
alert-community-2, alert-community-3, and  
page 41  
alert-community-4. You can configure these new  
parameters via the configuration files or the Aastra Web UI.  
Startup Enhancement  
480i CT Only  
Upon phone startup, the maximum time the phone spends  
attempting to contact the configuration server (TFTP, FTP or  
HTTP) has been reduced.  
page 45  
page 46  
Single Call Restriction (480i A new feature has been implemented on the 480i CT that  
CT only)  
allows an administrator to enable or disable a single call  
restriction between the 480i CT and a call server.  
9112i/9133i Only  
Addition of DisplayName1 & The 9112i and 9133i IP phones now support the parameters, page 48  
DisplayName2 (now also  
applies to 9112i/9133i)  
displayName1 and displayName2. Configurable via the  
configuration files and the Aastra Web UI.  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
SIP Timers  
The following paragraphs describe SIP timers that have been added or changed in  
Release 1.4.1.  
SIP Registration Renewal Timer  
A new parameter has been added to the IP phones that enables an administrator to  
control when registration renewals occur. The new parameter sip registration  
renewal timer specifies the length of time, in seconds, before the expiration of an  
existing registration, that the registration is renewed. For example, if the value is  
set to 20, then 20 seconds before the registration is due to expire, a new  
REGISTER message is sent to the registrar to renew the registration.  
The parameter may be set via the configuration files and the Aastra Web UI.  
Configuring via the Configuration Files  
You use the following parameter in the configuration files to control when the  
phone renews SIP registration:  
Parameter –  
sip registration renewal timer  
Aastra Web UI  
Advanced Settings->Global SIP->  
Advanced SIP Settings  
Configuration Files  
aastra.cfg, <mac>.cfg  
Registration Renewal Timer  
(in Web UI)  
Description  
Format  
The length of time, in seconds, that the phone renews registrations.  
Integer  
Default Value  
Range  
15  
0 to 214748364  
The value set for this parameter should be between 0 and the value set  
for the registration period.  
Example  
sip registration renewal timer: 10  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
Configuring via the Aastra Web UI  
You can set the Registration Renewal Timer in the Aastra Web UI at  
Advanced Settings->Global SIP->Advanced SIP Settings.  
Registration Renewal  
Timer  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
SIP Registration Timeout Retry Timer  
An administrator can now set the length of time, in seconds, that the phone waits  
until it re-attempts to register after a REGISTER message times out. This  
parameter is called, sip registration timeout retry timer, and is configurable via  
the configuration files and the Aastra Web UI.  
Configuring via the Configuration Files  
You use the following parameter in the configuration files to configure the  
registration timeout retry timer.  
Parameter –  
sip registration timeout retry  
timer  
Aastra Web UI  
Advanced Settings->Global SIP->  
Advanced SIP Settings  
aastra.cfg, <mac>.cfg  
Configuration Files  
Registration Timeout Retry  
Timer  
(in Web UI)  
Description  
Specifies the length of time, in seconds, that the phone waits until it  
re-attempts to register after a REGISTER message times out.  
Note: If this parameter is set lower than 30 seconds, the phone uses a  
minimum timer of 30 seconds.  
Format  
Integer  
Default Value  
Range  
120  
30 to 214748364  
Example  
sip registration timeout retry timer: 150  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
Configuring via the Aastra Web UI  
You can set “Registration Timeout Retry Timer” in the Aastra Web UI at  
Advanced Settings->Global SIP->Advanced SIP Settings.  
Registration Timeout  
Retry Timer  
“SIP Registration Retry Timer” Web UI Parameter Name Change  
In the Aastra Web UI only, the registration retry timer parameter name has  
been changed to “registration failed retry timer”. The functionality of this  
parameter has not changed. The default value is still 1800 seconds.  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Explicit Message Waiting Indicator (MWI) Subscription Period  
The IP phones support a new feature for MWI that allows the administrator to set  
a requested duration, in seconds, before an explicit MWI subscription times out.  
The phone re-subscribes to MWI before the subscription period ends.  
The new parameter you can configure is “sip explicit mwi subscription period”.  
You can configure this parameter using the configuration files or the Aastra Web  
UI.  
Configuring via the Configuration Files  
You configure the explicit MWI timeout using the following parameter in the  
configuration files.  
Parameter –  
sip explicit mwi subscription  
period  
Aastra Web UI  
Advanced Settings->Global SIP->  
Advanced SIP Settings  
Configuration Files aastra.cfg, <mac>.cfg  
Explicit MWI Timeout  
(in Web UI)  
Description  
The requested duration, in seconds, before the MWI subscription times  
out. The phone re-subscribes to MWI before the subscription period  
ends.  
Format  
Integer  
Default Value  
Range  
86400  
30 - 214748364  
sip explicit mwi timeout: 30  
Example  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
Configuring via the Aastra Web UI  
You configure “Explicit MWI Subscription Period” in the Aastra Web UI at  
Advanced Settings->Global SIP->Advanced SIP Settings.  
Explicit MWI Subscription  
Period  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Prefix Dialing  
The IP phones now support a prefix dialing feature for outgoing calls.  
How it works  
You can manually dial a number or dial a number from a list. The phone  
automatically maps the pre-configured prepended digit in the configuration, to the  
outgoing number. When a match is found, the prepended digits are added to the  
beginning of the dial string and the call is dialed.  
Note: The prepend digits are also added if the dialing times-out on a  
partial match.  
You can enable this feature by adding a prepend digit(s) to the end of the Local  
Dial Plan parameter string in the configuration files or the Aastra Web UI at  
Basic Settings->Preferences->General.  
For example, if you add a prepend map of “[2-9]XXXXXXXXX,91”, the IP  
phone adds the digits “91” to any 10-digit number beginning with any digit from 2  
to 9 that is dialed out. Other examples of prepend mappings are:  
1X+#,9 (Prepends 9 to any digit string beginning with “1” and terminated  
with “#”.)  
6XXX,579 (Prepends “579” to any 4-digit string starting with “6”.)  
[4-6]XXXXXX,78 (Prepends “78” to any 7-digit string starting with “4”, “5”, or “6”.)  
Note: You can configure a local dial plan via the configuration files or  
the Aastra Web UI.  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
Example  
If you enter the following dial string for a local dial plan:  
sip dial plan: 1+#,9  
where “9” is the prepended digit, and you dial the following number:  
15551212  
the IP phone automatically adds the “9” digit to the beginning of the dialed  
number before the number is forwarded as 915551212.  
Configuring via the Configuration Files  
You use the following parameter to configure local dial plan with a prepended  
digit(s).  
Parameter –  
sip dial plan  
Aastra Web UI  
Configuration Files  
Basic Settings->Preferences  
aastra.cfg, <mac>.cfg  
Local Dial Plan  
(in Web UI)  
Description  
A dial plan describes the number and pattern of digits that a user dials to  
reach a particular telephone number. The SIP local dial plan is as follows:  
Symbol  
Description  
0, 1, 2, 3, 4, 5, 6, 7, 8, 9  
Digit symbol  
X
Match any digit symbol (wildcard)  
Other keypad symbol  
Expression inclusive OR  
0 or more of the preceding digit symbol  
or [] expression  
Symbol inclusive OR  
Used only with [], represent a range of  
acceptable symbols; For example, [2-8]  
In the configuration files, enter the sip dial  
plan value using quotes.  
*, #, .  
|
+
[]  
-
“,” (open/close quotes)  
Note: You can configure prefix dialing by adding a prepend digit to the  
dial string. For example, if you add a prepend map of  
“[2-9]XXXXXXXXX,91”, the IP phone adds the digits “91” to any 10-digit  
number beginning with any digit from 2 to 9 that is dialed out. Other  
examples of prepend mappings are:  
1X+#,9 (  
6XXX,579 (Prepends “579” to any 4-digit string starting with “6”.)  
[4-6]XXXXXX,78 (Prepends “78” to any 7-digit string starting with “4”,  
“5”, or “6”.  
Format  
Alphanumeric characters  
X+#|XX+*  
Default Value  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Range  
Up to 127 alphanumeric characters  
sip dial plan: "1X+#,9”  
Example  
Configuring via the Aastra Web UI  
You configure a local dial plan string in the Aastra Web UI at  
Basic Settings->Preferences. The following illustration shows a Local Dial Plan  
dial string in the Aastra Web UI using a prepended digit.  
Local Dial Plan  
Last Number Redial  
The IP phones now have an enhanced redial user interface that allows a user  
to quickly redial the last number that was dialed out from the phone. You can:  
Press the REDIAL button twice to redial the last number dialed.  
Press the REDIAL button once, scroll the list of numbers, then press the  
REDIAL button again to dial the number that displays on the screen.  
This feature is static and is not configurable.  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
XML Objects and Enhancements  
The following paragraphs describe the XML enhancements in release 1.4.1.  
XML: AastraIPPhoneStatus object  
A new XML AastraIPPhoneStatus object has been implemented on the IP  
phones. This object provides the ability to display a status message on a single  
designated line on the phone’s idle screen when XML information is pushed from  
the servers.  
The 480i/480i CT phones display messages on the second line in the phone  
window. (where “No Service” would display if there was no service. If there is no  
service on the phone, the “No Service” message overrides the XML object  
message). The 9112i/9133i phones display messages on the first line (overriding  
the DisplayName). Long messages that are wider then the phone screen get  
truncated.  
If the phone receives multiple messages, the first message received displays first  
and the remaining messages scroll consecutively one at a time. Messages remain  
displayed until they are removed (by the server) or the phone reboots. The  
AastraIPPhoneStatus object feature is always enabled.  
Note: You can set the amount of time, in seconds, that a message displays  
to the phone before scrolling to the next message. For more information  
about this feature, see “Scroll Delay Option” on page 21.  
AastraIPPhoneStatus Structure  
The AastraIPPhoneStatus object describes the structure of the XML document  
that is used to send status messages to the phone. The basic structure of the  
AastraIPPhoneStatus object is:  
<AastraIPPhoneStatus>  
<Session>My session ID</Session>  
<Message index="Msg index">Message</Message>  
<! -- Additional status messages may be added under new Message tags-->  
<AastraIPPhoneStatus/>  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
The ”My Session ID” attribute must be unique to the application sending the  
XML object to the phone. The application generates the session ID, which could  
be a combination of letters and numbers. There is a maximum of one <Session>  
tag per PhoneStatus object, so the <Session> tag is optional.  
Examples  
Example 1: The following is an example of using the AastraIPPhoneStatus  
object:  
<AastraIPPhoneStatus>  
<Session>abc12345</Session>  
<Message index="3">Server side call forwarding disabled</Message>  
<AastraIPPhoneStatus/>  
In this example, the AastraIPPhoneStatus object sends the default behavior with  
the status message (i.e., the status message is added to the scroll list).  
Example 2: You can also use the AastraIPPhoneStatus object to remove status  
messages from the display, by setting an empty tag for the <Message index> tag.  
The following example removes the status message that was posted to the phone  
in Example 1.  
<AastraIPPhoneStatus>  
<Session>abc12345</Session>  
<Message index="3"/>  
<AastraIPPhoneStatus/>  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
Beep Option  
You can enable or disable a BEEP option in the AastraIPPhoneStatus object.  
When the phone receives a status message, the BEEP notifies the user that the  
message is being displayed. The following attribute enables/disables the BEEP  
from being heard:  
< AastraIPPhoneStatus Beep="yes|no">  
(case sensitive)  
This attribute is optional. If notification is required, the attribute must be in the  
ROOT. If the BEEP attribute is set to "yes" (i.e. Beep="yes") then it is an  
indication to the phone to sound a beep when it receives the object. If the Beep  
attribute is set to "no" (i.e. Beep="no") or not present, then the default behavior is  
no beep is heard when the object arrives to the phone.  
Beep Option via Configuration Files and Aastra Web UI  
The BEEP option can also be enabled or disabled via the configuration files and  
the Aastra Web UI using the following parameters:  
xml beep notification (via configuration files)  
XML Beep Support (via the Aastra Web UI)  
The value set in the configuration files and Aastra Web UI override the attribute  
you specify in the AastraIPPhoneStatus object.  
For example, if the AastraIPPhoneStatus object has the attribute of Beep=”yes”,  
and you uncheck (disable) the “XML Beep Support” in the Aastra Web UI, the  
phone does not beep when it receives an AastraIPPhoneStatus object.  
Setting the BEEP option in the configuration files and the Aastra Web UI is  
dynamic and applies to the phone immediately.  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Configuring via the Configuration Files  
You enable/disable the BEEP option using the following parameter in the  
configuration files:  
Parameter –  
Aastra Web UI  
Basic Settings->Preferences  
xml beep notification  
Configuration Files aastra.cfg, <mac>.cfg  
XML Beep Support  
(in Web UI)  
Description  
Enables or disables a BEEP notification on the phone when an  
AastraIPPhoneStatus object containing a “beep” attribute arrives to the  
phone.  
Format  
Boolean  
1 (ON)  
Default Value  
Range  
0 (OFF)No beep is audible even if the beep attribute is present in  
the XML object.  
1 (ON)The phone beeps when an XML object with the “beep”  
attribute arrives to the phone.  
Example  
xml beep notification: 0  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
Configuring via the Aastra Web UI  
You enable/disable the BEEP option in the Aastra Web UI at  
Basic Settings->Preferences.  
XML Beep Support  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Scroll Delay Option  
The IP phones support a scroll delay option that allows you to set the time delay,  
in seconds, between the scrolling of each status message on the phone. The default  
time is 5 seconds for each message to display before scrolling to the next message.  
You can configure this option via the configuration files or the Aastra Web UI.  
Changes are dynamic and apply to the phone immediately.  
Configuring via the Configuration Files  
You set a value for the scroll delay option using the following parameter in the  
configuration files.  
Parameter –  
Aastra Web UI  
Basic Settings->Preferences  
xml status scroll delay  
Configuration Files aastra.cfg, <mac>.cfg  
Status Scroll Delay (seconds)  
(in Web UI)  
Description  
Specifies the length of time, in seconds, that each XML status  
message displays on the phone.  
Format  
Integer  
Default Value  
Range  
5
1 to 25  
Example  
xml status scroll delay: 3  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
Configuring via the Aastra Web UI  
You set the scroll delay option in the Aastra Web UI at  
Basic Settings->Preferences.  
Status Scroll Delay  
(seconds)  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
XML: AastraIPPhoneExecute Object  
A new XML AastraIPPhoneExecute object has been implemented on the IP  
phones. This object provides the ability to execute commands on the phone using  
XML. Release 1.4.1 supports the following Execute object commands:  
Reset - This command waits until the phone is idle and then executes a reset.  
NoOp - This command has no affect on the IP phone. It is made up of a blank  
URI. You can use this feature when you need to press a key on the phone to  
access a feature, and it is not necessary to display anything.  
Since the server forces phone firmware changes, the AastraIPPhoneExecute  
object was implemented to send the reset command to the phone.  
AastraIPPhoneExecute Object Structure  
The AastraIPPhoneExecute object describes the structure of the XML document  
that is used to send a command to the phone. It delivers multiple execution  
requests to the phone. The basic structure of the AastraIPPhoneExecute object is:  
<AastraIPPhoneExecute>  
<ExecuteItem URI ="the URL or URI to be executed"/>  
<! -- Additional execution items may be added under new ExecuteItem tag-->  
</AastraIPPhoneExecute>  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
Using the Reset Command  
The <ExecuteItem URI =""/> tag can be entered with the command the phone  
should execute. Upon receiving an AastraIPPhoneExecute object, the phone  
begins executing the URL or URI specified.  
The following example shows an AastraIPPhoneExecute object using the Reset  
command:  
<AastraIPPhoneExecute>  
<ExecuteItem URI="Command: Reset"/>  
</AastraIPPhoneExecute>  
Note: If you specify a command as a URI attribute (instead of a URL),  
the keyword "Command" must be prepended in the value of the URI  
attribute so that the phone recognizes it as a URI attribute value. If you  
enter a URI and leave out the “Command” keyword, the phone interprets  
the value in the URI attribute as a URL containing network resources.  
The following example shows the AastraIPPhoneExecute object using a URL:  
<AastraIPPhoneExecute>  
<ExecuteItem URI="http://aastraserver/message.xml"/>  
</AastraIPPhoneExecute>  
When the phone receives this object, it displays the specified XML URI page.  
Using the NoOp Command  
You can use the AastraIPPhoneExecute object as an object to create a blank  
display (it has no affect on the IP phone). It is made up of a blank URI. You can  
use this feature when you need to press a key on the phone to access a feature, and  
it is not necessary to display anything.  
The following example shows an AastraIPPhoneExecute object using a blank  
URI:  
<AastraIPPhoneExecute>  
<ExecuteItem URI=””/>  
<AastraIPPhoneExecute>  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
XML: Action URI  
This feature provides the administrators the ability to specify a uniform resource  
identifier (URI) that triggers a GET when certain events occur. The IP phone  
events that support this feature are:  
Startup  
Successful registration  
Incoming call  
Outgoing call  
Offhook  
Onhook  
The following table identifies the configurable action URI parameters in the  
configuration files and the Aastra Web UI. This table also identifies the variables  
that apply to specific parameters.  
Configuration File Aastra Web UI Parameters at  
Applicable Variables  
Parameters  
Advanced Settings->Action URI  
action uri startup  
Startup  
-
action uri registered  
Successful Registration  
$$SIPUSERNAME$$  
$$SIPAUTHNAME$$  
$$PROXYURL$$  
action uri incoming  
Incoming Call  
$$REMOTENUMBER$$  
$$DISPLAYNAME$$  
$$SIPUSERNAME$$  
$$INCOMINGNAME$$  
action uri outgoing  
Outgoing Call  
$$REMOTENUMBER$$  
$$SIPUSERNAME$$  
action uri offhook  
action uri onhook  
Offhook  
Onhook  
-
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
How it works  
When a startup, successful registration, incoming call, outgoing call,  
offhook, or onhook call event occurs on the phone, the phone checks to see if  
the event has an action URI configured. If the phones finds a URI configured, any  
variables configured (in the form $$VARIABLENAME$$) are replaced with the  
value of the appropriate variable. After all of the variables are bound, the phone  
executes a GET on the URI. The Action URI binds all variables and is not  
dependant on the state of the phone.  
For example, if you enter the following string for the action uri outgoing  
parameter:  
action uri outgoing: http://10.50.10.140/  
outgoing.pl?number=$$REMOTENUMBER$$  
and you dial out the number 5551212, the phone executes a GET on:  
Note: If the phone can’t find the Action URI you specify, it returns a  
“NULL” response. For example,  
You can configure this feature via the configuration files or the Aastra Web UI.  
Configuring via the Configuration Files  
You use the following parameters in the configuration files to configure the XML:  
action URI.  
Parameter –  
action uri startup  
Aastra Web UI  
Configuration Files  
Advanced Settings->Action URI  
aastra.cfg, <mac>.cfg  
Startup  
(in Web UI)  
Description  
Specifies the URI for which the phone executes a GET on when a startup  
event occurs.  
Format  
Fully qualified URI  
Default Value  
Range  
Not Applicable  
Up to 128 ASCII characters  
Example  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Parameter –  
action uri registered  
Aastra Web UI  
Configuration Files  
Advanced Settings->Action URI  
aastra.cfg, <mac>.cfg  
Successful Registration  
(in Web UI)  
Description  
Specifies the URI for which the phone executes a GET on when a  
successful registration event occurs. This parameter can use the  
following variables:  
$$SIPUSERNAME$$  
$$SIPAUTHNAME$$  
$$PROXYURL$$  
Format  
Fully qualified URI  
Not Applicable  
Default Value  
Range  
Up to 128 ASCII characters  
Example  
name=$$SIPAUTHNAME$$  
Parameter –  
action uri incoming  
Aastra Web UI  
Configuration Files  
Advanced Settings->Action URI  
aastra.cfg, <mac>.cfg  
Incoming Call  
(in Web UI)  
Description  
Specifies the URI for which the phone executes a GET on when an  
incoming call event occurs. This parameter can use the following  
variables:  
$$REMOTENUMBER$$  
$$DISPLAYNAME$$  
$$SIPUSERNAME$$  
$$INCOMINGNAME$$  
Format  
Fully qualified URI  
Not Applicable  
Default Value  
Range  
Up to 128 ASCII characters  
Example  
incoming.php?number=$$REMOTENUMBER$$  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
Parameter –  
action uri outgoing  
Aastra Web UI  
Configuration Files  
Advanced Settings->Action URI  
aastra.cfg, <mac>.cfg  
Outgoing Call  
(in Web UI)  
Description  
Specifies the URI for which the phone executes a GET on when an  
outgoing call event occurs. This parameter can use the following  
variables:  
$$REMOTENUMBER$$  
$$SIPUSERNAME$$  
Format  
Fully qualified URI  
Not Applicable  
Default Value  
Range  
Up to 128 ASCII characters  
Example  
outgoing.php?number=$$REMOTENUMBER$$  
Parameter –  
action uri offhook  
Aastra Web UI  
Configuration Files  
Advanced Settings->Action URI  
aastra.cfg, <mac>.cfg  
Offhook  
(in Web UI)  
Description  
Specifies the URI for which the phone executes a GET on when an  
offhook event occurs.  
Note: The only supported use of the offhook action URI is to perform a  
GET that returns a NoOp Execute XML object. Although the phone  
correctly displays all other returned XML objects, the interaction with  
other phone features, such as speeddial, redial, etc., is undefined.  
Format  
Fully qualified URI  
Default Value  
Range  
Not Applicable  
Up to 128 ASCII characters  
Example  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Parameter –  
action uri onhook  
Aastra Web UI  
Configuration Files  
Advanced Settings->Action URI  
aastra.cfg, <mac>.cfg  
Onhook  
(in Web UI)  
Description  
Specifies the URI for which the phone executes a GET on when an  
onhook event occurs.  
Format  
Fully qualified URI  
Default Value  
Range  
Not Applicable  
Up to 128 ASCII characters  
Example  
Configuring via the Aastra Web UI  
You can set the XML: action parameters in the Aastra Web UI at  
Advanced Settings->Action URI.  
Action URI parameters  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
XML: Softkey URI  
In addition to specifying variables for the Action URIs, you can also specify  
variables in the XML softkey URIs that are bound when the key is pressed. These  
variables are the same as those used in the Action URIs.  
When an administrator enters an XML softkey URI either via the Aastra Web UI  
or the configuration files, they can specify the following variables:  
$$SIPUSERNAME$$  
$$SIPAUTHNAME$$  
$$PROXYURL$$  
$$REMOTENUMBER$$  
$$DISPLAYNAME$$  
$$INCOMINGNAME$$  
When the softkey is pressed, if the phone finds a URI configured with variables  
(in the form $$VARIABLENAME$$), they are replaced with the value of the  
appropriate variable. After all of the variables are bound, the softkey executes a  
GET on the URI.  
Example  
For example, if the administrator specifies an XML softkey with the value:  
http://10.50.10.140/script.pl?name=$$SIPUSERNAME$$  
This softkey executes a GET on:  
assuming that the sip username of the specific line is 42512.  
You can configure the XML softkey URI variables via the configuration files or  
the Aastra Web UI.  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Configuring via Configuration Files  
You use the following parameters to configure XML softkeys using variable  
binding:  
softkeyN type  
softkeyN label  
softkeyN value  
prgkeyN type  
prgkeyNvalue  
For example, on a 480i/480i CT:  
softkey1 type: xml  
softkey1 label: JohnSmith  
softkey1 value: http://10.50.10.140/script.pl?name=$$SIPUSERNAME$$  
On a 9112i/9133i:  
prgkey1 type: xml  
prgkey1 value: http://10.50.10.140/script.pl?name=$$SIPUSERNAME$$  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
Configuring via the Aastra Web UI  
For 480i/480i CT, you configure XML softkey variable binding at  
Operation->Softkeys and XML.  
For 9112i/9133i, you configure XML softkey variable binding at  
Operation->Programmable Keys.  
XML Softkey configured using variable binding  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
XML: HTTP Refresh Header  
A new HTTP refresh header feature has been implemented on the IP phones. This  
feature includes the following:  
All current XML screen objects have the ability to be refreshed by adding a  
Refresh and URL setting to the HTTP headers. (see Refresh setting format  
below)  
The Refresh setting is set by the XML application and it is up to the  
application to decide which objects it wants to refresh.  
Note: This HTTP refresh header feature only applies to objects that  
display to the screen.  
The Refresh setting must be included in the HTTP header. The XML application  
decides which objects it wants to use with this setting. The phone recognizes this  
setting when parsing the HTTP header. If the setting is present, then it passes  
along the refresh timeout and the URL to the ParserData object, which all XML  
screen objects inherit from. The ParserData class also has a timer, which must be  
set to expire at the next refresh time. When the timer expires (time to refresh the  
screen), the phone requests the URL again and displays the refreshed screen.  
Refresh Setting Format  
The following is the Refresh setting format for the HTTP header:  
Refresh: <timeout>; URL=<page to load>  
The following example is a Refresh setting for use in an HTTP header:  
Note: You must use the Refresh and URL parameters in order for this  
feature to work in the HTTP header.  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
Backup Proxy/Registrar Support  
The IP phones now support a backup SIP proxy and backup SIP registrar feature.  
If the primary server is unavailable, the phone automatically switches to the  
backup server allowing the user's phone to remain in service.  
How it Works  
All SIP registration messages are sent to the primary registrar first. If the server is  
unavailable, then a new registration request is sent to the backup registrar. This  
also applies to registration renewal messages, which try the primary server before  
the backup.  
Similarly, any outgoing calls attempt to use the primary proxy first, then the  
backup if necessary. In addition, subscriptions for BLF, BLA, and explicit MWI  
can also use the backup proxy when the primary fails. Outgoing calls and the  
previously mentioned subscriptions behave the same as registrations, where the  
primary proxy is tried before the backup.  
You can configure the backup SIP proxy on a global or per-line basis via the  
configuration files or the Aastra Web UI.  
Configuring via the Configuration Files  
You can configure the following parameters for a backup SIP proxy and backup  
SIP registrar.  
Global parameters  
Parameter –  
sip backup proxy ip  
Aastra Web UI  
Advanced Settings->Global SIP->  
Basic SIP Network Settings  
aastra.cfg, <mac>.cfg  
Configuration Files  
Backup Proxy Server  
(in Web UI)  
Description  
The IP address of the backup SIP proxy server for which the IP phone  
uses when the primary SIP proxy is unavailable.  
Format  
IP address or fully qualified Domain Name  
0.0.0.0  
Default Value  
Range  
Not Applicable  
Example  
sip backup proxy ip: 192.168.0.102  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Parameter –  
sip backup proxy port  
Aastra Web UI  
Advanced Settings->Global SIP->  
Basic SIP Network Settings  
aastra.cfg, <mac>.cfg  
Configuration Files  
Backup Proxy Port  
(in Web UI)  
Description  
Format  
The backup proxy’s port number.  
Integer  
Default Value  
Range  
0
Not Applicable  
Example  
sip backup proxy port: 5060  
Parameter –  
Aastra Web UI  
Advanced Settings->Global SIP->  
sip backup registrar ip  
Basic SIP Network Settings  
aastra.cfg, <mac>.cfg  
Configuration Files  
Backup Registrar Server  
(in Web UI)  
Description  
The address of the backup registrar (typically, the backup SIP proxy) for  
which the IP phone uses to send REGISTER requests if the primary  
registrar is unavailable.  
A global value of 0.0.0.0 disables backup registration. However, the  
phone is still active and you can dial using username@ip address of the  
phone.  
If the backup registrar IP address is set to 0.0.0.0 for a per-line basis (i.e,  
line 1, line 2, etc.), then the backup register request is not sent, the "No  
Service" message does not display, and the message waiting indicator  
(MWI) does not come on.  
Format  
IP address or fully qualified Domain Name  
0.0.0.0  
Default Value  
Range  
Not Applicable  
Example  
sip backup registrar ip: 192.168.0.102  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
Parameter –  
sip backup registrar port  
Aastra Web UI  
Advanced Settings->Global SIP->  
Basic SIP Network Settings  
aastra.cfg, <mac>.cfg  
Configuration Files  
Backup Registrar Port  
(in Web UI)  
Description  
Format  
The backup registrar's (typically the backup SIP proxy) port number.  
Integer  
Default Value  
Range  
0
Not Applicable  
Example  
sip backup registrar port: 5060  
Per-line parameters  
Parameter –  
sip linex backup proxy ip  
Aastra Web UI  
Advanced Settings->LineN->  
Basic SIP Network Settings  
aastra.cfg, <mac>.cfg  
Configuration Files  
Backup Proxy Server  
(in Web UI)  
Description  
The IP address of the backup SIP proxy server for which the IP phone  
uses when the primary SIP proxy is unavailable.  
Format  
IP address or fully qualified Domain Name  
0.0.0.0  
Default Value  
Range  
Not Applicable  
Example  
sip line1 backup proxy ip: 192.168.0.102  
Parameter –  
sip linex backup proxy port  
Aastra Web UI  
Advanced Settings->LineN->  
Basic SIP Network Settings  
aastra.cfg, <mac>.cfg  
Configuration Files  
Backup Proxy Port  
(in Web UI)  
Description  
Format  
The backup proxy’s port number.  
Integer  
Default Value  
Range  
0
Not Applicable  
Example  
sip line1 backup proxy port: 5060  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Parameter –  
sip linex backup registrar ip  
Aastra Web UI  
Advanced Settings->LineN->  
Basic SIP Network Settings  
aastra.cfg, <mac>.cfg  
Configuration Files  
Backup Registrar Server  
(in Web UI)  
Description  
The address of the backup registrar (typically, the backup SIP proxy) for  
which the IP phone uses to send REGISTER requests if the primary  
registrar is unavailable.  
A global value of 0.0.0.0 disables backup registration. However, the  
phone is still active and you can dial using username@ip address of the  
phone.  
If the backup registrar IP address is set to 0.0.0.0 for a per-line basis (i.e,  
line 1, line 2, etc.), then the backup register request is not sent, the "No  
Service" message does not display, and the message waiting indicator  
(MWI) does not come on.  
Format  
IP address or fully qualified Domain Name  
0.0.0.0  
Default Value  
Range  
Not Applicable  
Example  
sip line1 backup registrar ip: 192.168.0.102  
Parameter –  
sip linex backup registrar port  
Aastra Web UI  
Advanced Settings->LineN->  
Basic SIP Network Settings  
aastra.cfg, <mac>.cfg  
Configuration Files  
Backup Registrar Port  
(in Web UI)  
Description  
Format  
The backup registrar's (typically the backup SIP proxy) port number.  
Integer  
Default Value  
Range  
0
Not Applicable  
Example  
sip line1 backup registrar port: 5060  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
Configuring via the Aastra Web UI  
For global configuration, you can set the backup SIP proxy/registrar parameters at  
Advanced Settings->Global SIP->Basic SIP Network Settings.  
For per-line configuration, you can set the backup SIP proxy/registrar parameters  
at Advanced Settings->LineN->Basic SIP Network Settings.  
Backup SIP Proxy/  
Registrar Parameters  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Auto-discovery Using mDNS  
Release 1.4.1 introduces a process that allows the phones to auto-discover all  
servers on a network using mDNS. When the IP phone discovers a TFTP server, it  
is automatically configured by that TFTP server.  
An unconfigured phone (phone right out of the box) added to a network, attempts  
to auto-discover a configuration server on the network without any end-user  
intervention. When it receives DHCP option 66 (TFTP server), it automatically  
gets configured by the TFTP server.  
An already configured phone (either previously configured by auto-discovery or  
manually configured) added to a network, uses its predefined configuration to  
boot up.  
Notes:  
1. Configuration parameters received via DHCP do not constitute  
configuration information, with the exception of a TFTP server.  
Therefore, you can plug a phone into a DHCP environment, still use the  
auto-discovery process, and still allow the use of the TFTP server  
parameter to set the configuration server.  
2. DHCP option 66 (TFTP server details) overrides the mDNS phase of  
the auto-discovery. Therefore, the DHCP option takes priority and the  
remaining process of auto-discovery continues.  
3. As the phone performs auto-discovery, all servers in the network  
(including the TFTP server), display in the phone window. However, only  
the server configured for TFTP automatically configures the phone.  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
IP Phone Features for Sylantro Servers  
Last Call Return (lcr) Support  
A new feature has been added to the IP phones that allow a user or administrator  
to configure a "last call return" function on a softkey or programmable key.  
This feature is for Sylantro servers only.  
You can configure the “lcr” softkey feature via the configuration files and the  
Aastra Web UI.  
How it works  
If you configure "lcr" on a softkey or programmable key, and a call comes  
into your phone, after you are finished with the call and hang up, you can  
press the key configured for “lcr” and the phone dials the last call you received.  
When you configure an “lcr” softkey, the label “LCR” displays next to that  
softkey on the IP phone. When the Sylantro server detects an “lcr” request, it  
translates this request and routes the call to the last caller.  
For the 480i/480i CT, applicable parameters to configure for “lcr” are:  
softkeyN type (“Type” in Web UI)  
softkeyN line (“Line” in Web UI)  
softkeyN states (“Idle, Connected, Incoming, Outgoing” in Web UI)  
For the 9112i/9133i, applicable parameters to configure for “lcr” are:  
prgkeyN type (“Type” in Web UI)  
prgkeyN line (“Line” in Web UI)  
Note: For more information about “lcr” see the  
SIP IP Phone Administrator Guide, Release 1.4.1.  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Support for additional “Alert Info” keywords for distinctive ringing  
New configurable, distinctive ringing support has been added to release 1.4.1 for  
Sylantro servers. The following “info” parameters allow you to configure specific  
priority alert tones for each parameter that may appear as keywords in the  
“Alert-Info” header of a Sylantro server:  
alert-acd (auto call distribution)  
alert-community-1  
alert-community-2  
alert-community-3  
alert-community-4  
You can configure these priority alert parameters via the configuration files or the  
Aastra Web UI.  
Configuring via the Configuration Files  
In addition to the priority alert parameters that already exist (from previous  
releases), you can also now configure the following parameters for priority  
alerting.  
Parameter –  
alert auto call distribution  
Aastra Web UI  
Basic Settings->Preferences->  
Priority Alerting Settings  
aastra.cfg, <mac>.cfg  
Configuration Files  
auto call distribution  
(in Web UI)  
Description  
When an "alert-acd" keyword appears in the header of the INVITE  
request, the configured Bellcore ring tone is applied to the IP phone.  
Format  
Integer  
Default Value  
Range  
0
Normal ringing  
0
1
2
3
4
5
Normal ringing (default)  
Bellcore-dr2  
Bellcore-dr3  
Bellcore-dr4  
Bellcore-dr5  
Silent  
Example  
alert auto call distribution: 2  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
Parameter –  
alert community 1  
Aastra Web UI  
Basic Settings->Preferences->  
Priority Alerting Settings  
aastra.cfg, <mac>.cfg  
Configuration Files  
community-1  
(in Web UI)  
Description  
When an "alert community-1” keyword appears in the header of the  
INVITE request, the configured Bellcore ring tone is applied to the IP  
phone.  
Format  
Integer  
Default Value  
Range  
0
Normal ringing  
0
1
2
3
4
5
Normal ringing (default)  
Bellcore-dr2  
Bellcore-dr3  
Bellcore-dr4  
Bellcore-dr5  
Silent  
Example  
alert community 1: 3  
Parameter –  
alert community 2  
Aastra Web UI  
Basic Settings->Preferences->  
Priority Alerting Settings  
aastra.cfg, <mac>.cfg  
Configuration Files  
community-2  
(in Web UI)  
Description  
When an "alert community-2” keyword appears in the header of the  
INVITE request, the configured Bellcore ring tone is applied to the IP  
phone.  
Format  
Integer  
Default Value  
Range  
0
Normal ringing  
0
1
2
3
4
5
Normal ringing (default)  
Bellcore-dr2  
Bellcore-dr3  
Bellcore-dr4  
Bellcore-dr5  
Silent  
Example  
alert community 2: 4  
42  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Parameter –  
alert community 3  
Aastra Web UI:  
Basic Settings->Preferences->  
Priority Alerting Settings  
aastra.cfg, <mac>.cfg  
Configuration Files:  
community-3  
(in Web UI)  
Description  
When an "alert community-3” keyword appears in the header of the  
INVITE request, the configured Bellcore ring tone is applied to the IP  
phone.  
Format  
Integer  
Default Value  
Range  
0
Normal ringing  
0
1
2
3
4
5
Normal ringing (default)  
Bellcore-dr2  
Bellcore-dr3  
Bellcore-dr4  
Bellcore-dr5  
Silent  
Example  
alert community 3: 1  
Parameter –  
alert community 4  
Aastra Web UI:  
Basic Settings->Preferences->  
Priority Alerting Settings  
aastra.cfg, <mac>.cfg  
Configuration Files:  
community-4  
(in Web UI)  
Description  
When an "alert community-4” keyword appears in the header of the  
INVITE request, the configured Bellcore ring tone is applied to the IP  
phone.  
Format  
Integer  
Default Value  
Range  
0
Normal ringing  
0
1
2
3
4
5
Normal ringing (default)  
Bellcore-dr2  
Bellcore-dr3  
Bellcore-dr4  
Bellcore-dr5  
Silent  
Example  
alert community 4: 2  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
Configuring via the Aastra Web UI  
You can configure the new parameters for priority alerting in the Aastra Web UI at  
Advanced Settings->Action URI.  
New priority alert parameters  
44  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Startup Enhancement  
During startup, the maximum time the phone spends attempting to contact the  
configuration server (TFTP, FTP or HTTP) has been reduced from 10 minutes to  
approximately 30 seconds.  
When the phone boots up, it attempts to contact the configuration server (if the  
configuration server parameters are configured on the phone), for 4 seconds. The  
phone then displays a “Skip” button. After an additional 30 seconds has expired,  
or if the user presses the “Skip” button, the phone continues the boot process  
using the stored configuration.  
At any point during this period, if the phone successfully contacts the  
configuration server, it automatically continues and attempts to download its  
configuration and any new firmware from that server.  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
480i CT Only  
Single Call Restriction (480i CT only)  
A new feature has been implemented on the 480i CT that allows an administrator  
to enable or disable a single call restriction between the 480i CT base unit and a  
call server.  
When this feature is enabled (set to 1), you can make separate active calls from  
the 480i CT base unit and from the cordless handset. If this feature is disabled (set  
to 0), only one call can be active at a time either from the base unit or from the  
handset. When this feature is disabled, and you make an active call on either the  
base unit or the handset, any other attempt to make an active call is put on hold.  
Also, when this feature is disabled, more than one call can negotiate complex  
audio codecs since only a single call is decoding audio at a time.  
You can configure this feature via the configuration files or the Aastra Web UI.  
Configuring via the Configuration Files  
You can configure the following parameter for the 480i CT single media path  
restriction.  
Parameter –  
two call support  
Aastra Web UI  
Configuration Files  
Advanced Settings->Global SIP->RTP Settings  
aastra.cfg, <mac>.cfg  
Two Call Support  
(in Web UI)  
Description  
Enables or disables the single media path restriction between the  
480i CT base unit and the handset.  
When this feature is enabled (set to 1), you can make separate active  
calls from the 480i CT base unit and from the cordless handset. If this  
feature is disabled (set to 0), only one call can be active at a time either  
from the base unit or from the handset.  
When this feature is disabled, and you make an active call on either the  
base unit or the handset, any other attempt to make an active call is put  
on hold. Also, when this feature is disabled, more than one call can  
negotiate complex audio codecs since only a single call is decoding  
audio at a time.  
Format  
Boolean  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Default Value  
Range  
1
0 - Disable  
1 - Enable  
Example  
two call support: 0  
Configuring via the Aastra Web UI  
You can enable or disable the 480i CT single media path restriction at Advanced  
Settings->Global SIP->RTP Settings.  
Two Call Support  
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IP Phone Release Notes 1.4.1  
New Features in 1.4.1  
9112i/9133i Only  
Addition of DisplayName1 & DisplayName2 (now also applies to 9112i/  
9133i)  
The 9112i/9133i IP phones now support the use of displayName1 and  
displayName2. The value in these fields display on the idle screen of the IP  
phone. Previous IP phone releases supported these parameters on the  
480i/480i CT only.  
You can configure these parameters via the configuration files or the Aastra Web  
UI.  
Configuring via the Configuration Files  
You use the following parameters in the configuration files to configure  
displayName1 & displayName2.)  
Parameter –  
displayName1  
Aastra Web UI  
Configuration Files  
Basic Settings->Preferences  
aastra.cfg, <mac>.cfg  
Idle Display Name 1  
(in Web UI)  
Description  
The name displayed on the idle screen rather than the screen name and  
phone number  
Format  
Alphanumeric characters  
Not Applicable  
Default Value  
Range  
For 480i/480i CT:  
Up to 21 characters (width of LCD)  
For 9112i/9133i:  
Up to 16 characters (width of LCD)  
Example  
displayName1: SIPphone1  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
New Features in 1.4.1  
Parameter –  
displayName2  
Aastra Web UI  
Configuration Files  
Basic Settings->Preferences  
aastra.cfg, <mac>.cfg  
Idle Display Name 2  
(in Web UI)  
Description  
The name displayed on the idle screen rather than the screen name and  
phone number  
Format  
Alphanumeric characters  
Not Applicable  
Default Value  
Range  
For 480i/480i CT:  
Up to 21 characters (width of LCD)  
For 9112i/9133i:  
Up to 16 characters (width of LCD)  
Example  
displayName2: SIPphone2  
Configuring via the Aastra Web UI  
On the 9112i/9133i, you can configure Idle Display Name 1& Idle Display Name  
2 in the Aastra Web UI at Basic Settings->Preferences->General.  
Idle Display Name 1  
Idle Display Name 2  
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IP Phone Release Notes 1.4.1  
Issues Resolved in Release 1.4.1  
Issues Resolved in Release 1.4.1  
Description  
This section describes the issues resolved in release 1.4.1. 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 Number  
Description of Fix  
CLN04606  
With the 480i, the phone no longer hangs on startup when global call park and call  
pickup keys are configured.  
CLN04933  
DEF04090  
DEF04096  
DEF04174  
The IP phones no longer crash during transfers using BroadWorks release 1.2.  
Phone no longer adds extra “/” in XML requests.  
The cordless handsets now support sending DTMF events as INFO packets.  
The message “No Service” no longer appears momentarily on the 480i during  
registration. A parameter “sip registration renewal timer” has been added that allows  
you to set the time, in seconds, prior to the expiration of when the registration is  
renewed.  
DEF04175  
DEF04178  
DEF04188  
Ringing is no longer interrupted by the BLF ringing alert tone.  
Added support for URI dialing to the 9133i programmable keys.  
The Calls between the cordless handset and the base now get a voice path when  
the call is not made using the intercom function.  
DEF04210  
DEF04234  
When the media port is equal to 0 in the INVITE, the phone no longer sends “486  
Busy Here” after sending “180 Ringing when using a MetaSwtich.  
The IP phones no longer reject calls when receiving an attribute in SDP that it didn’t  
recognize. It now correctly ignores those attributes. This fixes an issue with the  
Eyebeam soft client.  
DEF04302  
DEF04450  
DEF04545  
You can now toggle to handsfree while on hold.  
When using the Aastra Web UI, the dial plan no longer truncates at 85 characters.  
The hangup key now returns dialtone when the handset stays off hook. With the 480i  
CT, the y  
key now acts consistently across all phones.  
50  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
Issues Resolved in Release 1.4.1  
Issue Number  
Description of Fix  
DEF04757  
The call transfer (Xfer) function was not working with Broadsoft SCA using multiple  
headers. The call transfer now works on an SCA line with Broadsoft R13.  
DEF05117  
If the CODEC negotiation fails during the processing of a re-INVITE, the phone no  
longer drops the call, but instead responds with a “488 Not Acceptable Here”  
message.  
Enhancements/Changes  
This section describes the enhancements and changes made to the 1.4.1 release..  
Note: Unless specifically indicated, these enhancements apply to all  
phone models.  
Issue Number  
Description of Fix  
DEF04638  
ENH04358  
XML URIs now support adding a port number to the URI.  
Added ability to change debug level from Aastra Web UI. You can now configure a  
“log level” in the Troubleshooting section of the Web interface.  
ENH04371  
Web recovery now resets the administrator password and removes local  
configuration files. You force the phone into recovery mode by holding ‘#’ and ‘1’ on  
startup.  
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IP Phone Release Notes 1.4.1  
Known Anomalies in 1.4.1  
Known Anomalies in 1.4.1  
Description  
This section describes the known anomalies in release 1.4.1.  
Note: Unless specifically indicated, these known anomalies apply to all  
phone models.  
Issue Number  
Description  
DEF04448  
DEF04535  
DEF04899  
DEF04912  
DEF05019  
SIP contact URI is not used in subsequent REFER-TO address.  
The TO field displays your own name during an originated call.  
In XML APIs, password is not hidden when it is marked as not editable.  
Sending a lot of messages (hundreds) within a second, causes the phone to lock up.  
The previous IP Phone release 1.4 shows the softkeys for the XML  
AastraIPPhoneInputScreen object as:  
1 = Backspace  
2 = Dot  
3 = ChangeCase  
4 = Numeric/Alpha  
5 = Cancel  
6 = Done  
This is incorrect. For softkeys 5 and 6, the key assignment is:  
5 = Done  
6 = Cancel  
DEF05056  
ENH05218  
An incoming intercom call changes the current audio mode (speaker/headset) of  
existing calls.  
When dialing a number on the phone and another call comes in, the phone rings and  
the digits you dialed are lost.  
52  
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SIP IP Phone Models 480i, 480i CT, 9112i, 9133i Release Notes  
Contacting Aastra Telecom Support  
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  
41-001024-00, Rev 09, Release 1.4.1  
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Generic SIP IP Phone  
Models 4801, 480i CT, 9112i, 9133i  
1.4.1 Release Notes  
Copyright © 2005-2006 Aastra Telecom. All rights reserved. Information in this document is  
subject to change without notice. Aastra Telecom assumes no responsibility for any errors that  
may appear in this document. Product capabilities described in this document pertain solely to  
Aastra Telecom's marketing activities in the U.S. and Canada. Availability in other markets may  
vary.  
41-001024-00 Rev 09  
SIP IP Phones Release 1.4.1  
November 2006  
Download from Www.Somanuals.com. All Manuals Search And Download.  

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