These release notes apply to version 3.3.1F (and earlier) of the Polycom UC Software that
runs on SoundPoint IP, SoundStation IP, and VVX phones. For more information, refer to
the documents listed in Section 4.
1.1 Important Notes
1. This patch release resolves a field reported security issue. SoundPoint IP and
Sound Station IP phones may be vulnerable to Denial of Service attacks when
used in certain conditions. Sending HTTP GET requests with a broken
authorization header can produce a device restart under some circumstances
in certain models of phones.
For details, refer to Technical Bulletin TB66743 for details. The technical
bulletin can be downloaded from:
2. The configuration files, their respective parameters and defaults, as well as
the provisioning methods have been simplified but extensively modified
compared to previous releases. SOME OF THESE CHANGES ARE NOT
BACKWARD COMPATIBLE with configuration parameters from previous
software releases.
Before installing the software, it is highly recommended that you first
familiarize yourself with the changes outlined in the “Administrator‟s Guide for
the Polycom® UC Software – 3.3.0” and Technical Bulletin 60519 “Simplified
Configuration Improvements in Polycom® UC Software 3.3.0”.
See Section 4 for details on how to access these documents.
3. VVX 1500 products running release SIP 3.2.2 or later CANNOT BE
DOWNGRADED TO EARLIER SIP SOFTWARE OR BOOTROM SOFTWARE.
4. Upgrading VVX 1500 products to release SIP 3.2.2 or later require a more complex
procedure than is typical. This procedure is documented in technical bulletin
TB53522. Please consult this document before starting the upgrade.
5. This release does not include support for the SoundPoint IP 300, 301, 430, 500, 501, 600, 601 and SoundStation IP 4000 products. These products are termed
„Legacy Products‟ and will be supported for critical issue fixes on the SIP 2.1.x
release (IP 300, 500), SIP 3.2.x (IP 430) and SIP 3.1.x release (for the other Legacy
models). Technical Bulletin TB35311 describes how to support these Legacy models
in an environment where SIP 3.2.0 or later is deployed for other phones. This
bulletin may be downloaded from:
http://www.polycom.com/support/voice/soundpoint_ip/VoIP_Technical_Bulletins_pub
.html. The template 000000000000.cfg file included with this release is set up to
6. SoundStation IP 7000/HDX Integration:
Release UCS 3.3.1 with BootROM 4.3.0 is recommended for SoundStation IP 7000
integration with Polycom HDX 4000/6000/7000/8000/9000 video systems running
one of the following releases:
The distribution of the SoundPoint / SoundStation IP / VVX SIP application UC Software
3.3.1F is done using two methods. Select the downloadable zip file(s) appropriate for your
deployment model.
In some cases it may be beneficial to download both release files. If this is necessary,
download both zip files, extract all the files from the „individual‟ release and then extract the
sip.ld file from the „combined‟ release file. All files other than “.ld” files are duplicated
between the two release zip files.
For centrally provisioned systems, download the appropriate file and extract the files to the
provisioning/boot server, maintaining the folder hierarchy present in the zip file.
Some of the configuration files must be modified. Refer to the documents listed in Section 4
for details.
The current build ID for all of the “.sip.ld” files llsted below (both split can combined) is now
at revision: 3.3.1. 0933
1.4.1 Release using individual (split) files
Use of „individual files‟ is recommended as it will result in a faster upgrade time for the
phone.
This method requires that all phones be running BootROM release 4.0.0 or later.
Configuration parameters: VoIP server/softswitch registration. Basic
settings
sip-interop.cfg
Configuration parameters: VoIP server/softswitch registration/interoperability configuration settings/registration
site.cfg
Configuration parameters: parameters expected to be set on a per-site
basis
video.cfg
Configuration parameters: Video connectivity
video-integration.cfg
Configuration parameters: For SoundStation IP7000/HDX Integration
SoundPointIP-dictionary.xml
Dictionary files for multilingual support include:
Chinese, China (for IP 450, 550, 560, 650 and IP 5000, 6000, 7000
only)
Danish, Denmark
Dutch, Netherlands
English, Canada
English, United Kingdom
English, United States
French, France
German, Germany
Italian, Italy
Japanese, Japan (for IP 450, 550, 560, 650, 670 and IP 5000, 6000,
7000 only)
Korean, Korea (for IP 450, 550, 560, 650, 670 and IP 5000, 6000,
7000 only)
Norwegian, Norway
Polish, Poland
Portuguese, Portugal
Russian, Russia
Slovenian, Slovenia
Spanish, Spain
Swedish, Sweden
SoundPointIPWelcome.wav
Start up welcome sound effect
LoudRing.wav
Loud ringer sound effect
Warble.wav
Loud ringer sound effect
Files
Description
sip.ld
Concatenated SIP application executable
sip.ver
Text file detailing build-id(s) for the release
000000000000.cfg
Example master configuration file
000000000000-directory~.xml
Example per-phone local contact directory XML file (edit and then
remove „~‟ from name to seed phones which have no directory)
The „combined‟ sip.ld file contains images for all members of the SoundPoint
IP/SoundStation IP/VVX products. This file is required for any phones that may be running
a BootROM release previous to SIP 4.0.0 (e.g. BootROM 3.2.3RevB).
Configuration parameters: VoIP server/softswitch registration. Basic
settings
sip-interop.cfg
Configuration parameters: VoIP server/softswitch registration/interoperability configuration settings/registration
site.cfg
Configuration parameters: parameters expected to be set on a per-site
basis
video.cfg
Configuration parameters: Video connectivity
video-integration.cfg
Configuration parameters: For SoundStation IP7000/HDX Integration
SoundPointIP-dictionary.xml
Dictionary files for multilingual support include:
Chinese, China (for IP 450, 550, 560, 650 and IP 5000, 6000, 7000
only)
Danish, Denmark
Dutch, Netherlands
English, Canada
English, United Kingdom
English, United States
French, France
German, Germany
Italian, Italy
Japanese, Japan (for IP 450, 550, 560, 650, 670 and IP 5000, 6000,
7000 only)
Korean, Korea (for IP 450, 550, 560, 650, 670 and IP 5000, 6000,
7000 only)
Norwegian, Norway
Polish, Poland
Portuguese, Portugal
Russian, Russia
Slovenian, Slovenia
Spanish, Spain
Swedish, Sweden
66743: Phones may be vulnerable to Denial of Service attacks when used in
certain configurations. Sending HTTP GET requests with a broken
authorization header can produce a device restart under certain
circumstances in certain models of phones. For full details, refer to Technical
Bulletin TB66743. See Section 4 Reference Documents for the location of the
documents.
2.2 Version 3.3.1
2.2.1 Added or Changed Features
52476: Added support for Premium extensions to server synchronized ACD
feature.
55059: Added support for Feature Key Synchronization using FAC/NOTIFY
message combination. Hosted IP solutions are implementing Synchronization
of Feature key Functions (e.g. DND/CFWD) using a Feature Access Code (FAC)
to set the Feature, and a SIP NOTIFY message to inform the phone of the
feature state.
55061: Added support for the Team Function feature. This feature extents the
compatibility of statically configured Busy Lamp Field (BLF) to operate in a
system requires the use of two URIs: one for call operations and another one
to subscribe for notification of dialog events. It also provides Ringing
Indication and a Directed call pick-up capability in a system that does not
generate RFC 4235 compliant dialog-info+xml documents.
50065: VVX 1500: Added support for CMA presence.
58888: Added the ability to trigger a reboot (or configuration update) from the
microbrowser. E.g. <softkey index="3" label="Reboot"
action="Action:UpdateConfig" />
59000: Phones now ignore BLA dialog documents (via NOTIFY) that are
reflected to User Agents that are party to the dialog.
60306: The server certificate Serial Number SN is now verified against the
server/proxy's „A record‟ domain names if the „SRV record‟ domain does not
match the SN.
61343: Phones now provide a configurable parameter that allows the
verification of the authentication tag to be disabled for received SRTP packets.
The purpose of this is to allow system administrators to resolve defects in
other endpoints where the authentication tag is not computed correctly.
Supported parameter: “sec.srtp.noAuthRxRTP”
61389: [802.1x - EAPOL Logoff] Phone will recycle the LAN link (e.g. it will
bring it down and up in an interval of one second) upon detecting a PC link
down event. This shall force the 802.1X switch to refresh the authorized port
state and start to send "request for identity" challenge messages. The
associated configuration parameter is:
“sec.dot1x.eapollogoff.pcforcelanlinkreset” with values:
"0" - Never recycle LAN link
"1" - Phone will unconditionally recycle the LAN link upon detecting PC
link down event
61861: Corporate Directory LDAP initialization supports the “bind”
authentication.
62115: SoundPoint IP 320, 321, 330, 331, 335: Phones now display the full text
strings of the “Phone Lock” feature.
62259: Phones now display the Call Forward destination on Idle Display.
62775: VVX 1500: The toolbar slide-out option is now configurable. The
associated configuration parameter is: “mb.main.toolbar.autohide.feature”
“1” - feature is enabled (default)
“0” - feature is disabled. The “Autohide” enable/disable buttons are no longer
visible to the user in the toolbar.
2.2.2 Removed Features
None
2.2.3 Corrections
44337: Configured characters ";", "/", "?", "&", "=", "~", "%", "\" are not
escaped (they are present) in INVITE messages.
55794: SoundStation IP 6000, 7000: Conference phone reboots upon receiving
a call with incorrect SRTCP indices.
56491: SIP 3.2.x: The screen displays the IP address of the server when
disabling the “Call Forwarding” feature using a „#‟ code. The screen displays
“21@ip_address_of_server” when it should display just “21”.
59824: Phone does not change all of the menu option labels into the selected
61090: The configuration parameter “voIpProt.SIP.musicOnHold.uri” is not
updated upon a configuration change.
61095: VVX 1500: While dialing a URL using the on-screen keyboard, the first
entered character is unexpectedly deleted.
61102: SoundStation IP 5000, 6000: The “Handset” or “Speaker” icon appears
(instead of the “Ringer” icon) when you adjust the ringer volume while the
phone is idle.
61104: VVX 1500: With a shared line configured on the phone, activity on the
remote shared line will cause the idle browser content to cycle off then on.
61114
: VVX 1500: Phone fails to boot-up with the DHCP VLAN 256 DVD option.
The user interface halts at the BootROM count-down screen, and fails to
respond to further key presses.
61115: Cannot answer calls for a few seconds after a configuration update is
invoked.
61242: The configuration parameter
“voIpProt.SIP.useCompleteUriForRetrieve” does not update upon a
configuration change.
61246: The “voIpProt.SIP.allowTransferOnProceeding” XML schema lists as
type=BOOL in the administrator‟s guide. The actual values are: 0, 1, & 2.
61273: Joining calls into local conference when 1 leg is a remotely held BLA
line results in no audio between both remote users.
61314: The number of characters for custom names is limited to 12. The
number has been extended to 127.
61367: When dialing a number with a „+‟ sign, e.g. +492101099210,
“user=phone” is not added to the “To” header.
61677: VVX 1500: The phone escapes the „%‟ character as „%25‟ when it is
present in the destination of a call.
61723: VVX 1500: The phone is missing the first string "<?xml version="1.0"
encoding="utf-8" ?>" in FAST UPDATE request which causes an integrated
RMX to reject the INFO method.
61779: Under certain conditions, the phone may reboot spontaneously from
idle state or in-use state.
61904: VVX 1500: A call is placed with the incorrect signaling protocol when
the line is configured as “dual line” protocol.
62036: SoundPoint IP 320,330: Phone stops sending DTMF RTP EVENTS when
receiving a second incoming call during an active primary call.
62114: VVX 1500: User cannot unlock the phone after the phone is locked with
a password containing letters.
62325: VVX 1500: Chinese characters cause the phone to become
remote party is held while reorder tone is played locally.
62490: Enabling the “Screen capture” function with httpd.enabled="0" causes
the phone to freeze and reboot.
62576: The phone does not reboot in order to pick up new “sip.ld” file after an
“Update Configuration” is invoked from the menu.
62642: The phone plays dial tone and RTP media when resuming on a call held
at another phone.
62704: BLA presence does not recover properly on the monitoring phone
when the LAN cable is disconnected and then re-connected.
62906: Phones do not correctly provision using the HTTPS protocol option
when using a server certificate with an older MD2 digest message algorithm.
63076: Phones with BLA lines are not able to establish more than 10 outgoing
calls.
63214: Phone will reboot if it receives more REFERs that
“reg.x.callsPerLineKey” is configured for.
2.2.4 Configuration Parameter Changes
The following table only lists the changes in Configuration Parameters when compared to
UCS 3.3.0.
The configuration files, their respective parameters and defaults, as well as the provisioning
methods have been simplified but extensively modified starting from UCS 3.3.0.
Before installing the software, it is highly recommended that you first familiarize yourself
with the changes outlined in the “Administrator‟s Guide for the Polycom® UC Software –
3.3.0” and Technical Bulletin 60519 “Simplified Configuration Improvements in Polycom®
UC Software 3.3.0”.
See Section 4 for details on how to access these documents.
23335: Configuration parameter values can now be updated at run-time.
24111: Improved the user interface for selecting a distinctive ring tone
associated with a contact in the local directory. You can now review the ring
name and play the ring tone before accepting and associating the ring tone for
specific contacts.
23394: Configuring parameters are now self-contained (default parameter
values) and the configuration process is more fault-tolerant.
35245: Line key behavior (configurable) has changed such that keys can now
be used to hang-up/terminate calls as well as establishing calls. The
associated configuration parameter is: param name=
"up.lineKeyCallTerminate" type="Bool" default="0" min="0" max="1".
38826: Added configuration parameters to expand the range of ports as well
as to randomize port selection for the purpose of downloading configuration
files to the phone using TCP connections.
support for dynamic support of G.729AB and iLBC codecs. G.729AB / iLBC.
48526: Simplified selection of codec configuration preferences. See TB60519
for details. THIS CHANGE IS NOT BACKWARD COMPATIBLE to configuration
files used with previous software releases.
48690: Phone Lock Feature: Added the ability for users to lock the phone and
restrict its access from unauthorized users. Users must enter a PIN in order to
access and use the phone. Refer to Quick Tip “QT 57215 Phone Lock Feature”
for additional information regarding this features use and configuration.
49658: Added configuration parameter to allow the phone to obtain “Caller ID”
from the “from” header instead of the “P-Asserted-Identity” segment. The
associated configuration parameter is: voIpProt.SIP.CID.sourcePreference =
“P-Asserted-Identity”, “Remote-Party-ID”, or “From”.
50067: Local contact directory now matches the Polycom CMA product‟s style
and user experience.
50151: Removed redundant levels of abstraction associated with arrays in
“config” files.
50644: VVX 1500: Improved the visual indicator of incoming calls for the
hearing impaired. Upon receiving an incoming call, the phone will ring and the
display will flash on and off with a bright orange and white screen. This visual
indicator can be seen even when the display is viewed at an indirect angle.
The associated configuration parameter is: up.accessibilityFeatures=”1”.
51121: RAM disk configuration parameters have been optimized.
51314: Added a configuration option to allow for minimal latency in order to
The associated configuration parameter is: voice.txPacketDelay.
“normal” or NULL (default) = no change to Tx latency; “low” = low delay
51523: Added the ability to scroll horizontally caller ID information (if it is
truncated when the number of characters in the caller ID string exceeds the
capacity of the display).
51446: Added configuration parameters supporting TLS cipher suites.
51594: Digit map replacements are not longer reflected in the placed calls list.
51725: VVX 1500: Added support for G.719 audio codec in H.323 calls.
51979: Added support for asymmetric audio codecs.
52253: Configuration parameter values modified by an “administrator” logon
credential using the phone‟s web server are not permitted to be altered by
“user” level logon credentials.
52459: Make website use the new configuration system.
52493: Added support for MD4 encryption key (OpenSSL).
52532: Phones no longer invoke a reboot during the uploading of override files
as a result of an unresponsive provisioning server (after a timeout).
52864: SoundPoint IP 320, 321, 330, 331, 335: Improved the user experience of
confirming a Local Directory Search.
53021: [CMA] Added support for NTLM version 2 authentication (via XMPP,
LDAP and HTTP(s)) for use with CMA.
53023: VVX 1500: Edit fields have been expanded to display additional
content.
53231: Added a configuration parameter to control the behavior of terminating
a 3-way conference by the conference initiator. Options now include either
terminating all conference legs or allowing the other parties to stay connected.
The associated configuration parameter is: “call.transferOnConferenceEnd”.
The default value is “1”. If set to “0”, then there is no transfer when a 3-way
conference is ended.
53417: VVX 1500: Implemented a slider bar for adjusting levels in various
menu screens.
53703: Added the ability for phones to send an 802.1x EAPOL Logoff message
on behalf of an attached PC when the PC is disconnected from the data port.
53932: Presence and BLF is supported on Avaya CS2100 soft switches.
54037: The method of attempting a Transfer / Conf of held party while in active
call is now consistent with all phones.
54045: Registration parameters can now be modified and activated without
requiring the phone to restart or reboot.
54098: Added the ability to automatically upgrade the BootBlock section of the
54167: The BootROM and application software versions may now be obtained
by using the on-board Web interface.
54301: A timestamp is displayed in Call Lists alongside the Caller ID.
54308: SoundPoint IP 320, 321, 330, 331, 335: The navigation keys can now be
used as a “spin box” control (ability to select values using the up and down
arrow keys) for numeric fields.
54678: Phones can now be deployed with a pre-set language. This supports
out-of-the box localization.
54928: Added a new API Telephony Event (XML) which is sent to the attached
application upon a successful line registration with a PBX.
55028: The maximum size of the contact directory contact field has been
increased to 128 to accommodate complex dialing scenarios.
55040, 57981: Added the ability for administrators to install custom device
certificates. The administrator can add private and public keys (certificate) via
TLS links.
55068: Added support for Null Ciphers to be used with TLS Authentication.
55318: The Advanced LDAP Search screen now supports languages other
than English.
55334: VVX 1500: Added the ability for the tool bar to hide automatically.
55490: The configuration Web interface has been expanded to include
parameters associated with security.
55508: When a “precedence” call is offered to the phone, it now rings with a
corresponding “precedence” ring tone.
55509: When a “precedence” outgoing call is initiated, a “precedence” style
ring-back tone is generated.
55510: The DSCP Differentiated Services Code Point levels for standard and
“precedence” level calls are aligned.
55513: The current “precedence” level of a call is displayed.
55546: The following diacritic letters and ligature are now supported (language
option selection) and can be displayed without having to change the character
encoding scheme: ä, ö, ü / Ä, Ö, Ü ß
55745: Phones now generate a MLPP resource-priority Header based on the
dialed number.
55985: SoundPoint IP 7000: Displays the "LogOut" soft key when configured
to be enabled.
56666, 56668: Added dynamic codec switching.
56790: Improved the computation of jitter buffer parameters based on received
Quality of Service QoS and expected payload size values.
56944: Improved the ability for application developers to implement changes
to the phone‟s configuration. Configuration parameters can be modified via
the web interface. The improved method also eliminates the need to reboot the
phone in order to register the changes.
57504: A new “Warble.wav” file is available which can be configured as an
audible ringer for incoming calls. This file will generate a loud ringer tone for
phones deployed in areas with a high ambient noise background.
58103: VVX 1500: The default maximum call data rate has been changed to 768
kbps. Change default maximum call rate to 768 kbps (from 512 kbps).
58156: VVX 1500: The user video call rate setting parameter value options
have been changed. Refer to the Administrator‟s Guide for details.
58758: VVX 1500: Improve the rendering performance of the browser.
58764: Added the ability of uploading configuration files representing the
phones' current set of configured parameter values to the provisioning server.
59307: Added a diagnostic menu option that enables the display of
configuration file statistics.
60316: Added an option in the user interface that allows the user to invoke the
phone to force it to re-configure itself based on newly administered
configuration file parameter values.
60353: Custom ring classes (se.rt) can now be set to a maximum value of 17.
60363: Custom ringer chords (tone.chord.ringer.spareX) can now be set to a
maximum value of 19.
2.3.2 Removed Features
50200, 53590: Removed configuration parameters that are no longer required
for custom bit-mapped graphic indicator icons.
56209: Removed support for the SoundPoint IP 430.
59917: Removed support for the animated idle display images (static idle
display images are still supported).
2.3.3 Corrections
33425: SoundStation IP 7000: Users could not reply to instant messages.
42509: VVX 1500: Cannot invoke speed dial list using the ArrowUP key when
first call is kept on hold.
43660: SoundPoint IP330: URL addresses are not saved in call list entry. When
the phone receives a URL call from SPIP@xxx.xxx.xxx.xxx, the phone does not
save the incoming URL call address into call list entry.
44034: SoundPoint IP 330: Cursor does not blink in hot dial prompt.
44278: The phone number is not displayed correctly on a line key when the
number of digits exceeds 10.
44478: VVX 1500: Configurable soft key features do not work.
44889: SoundPoint IP 330: The Polycom bitmapped logo is not displayed on
the phone‟s idle screen.
45013: Phones reboot after a “check-sync” request when a call is held and a
new call is initiated and then cancelled.
47135: VVX 1500: Casing of current encoding indication at title bar should
match corresponding soft keys.
47542: VVX 1500: The URL entry field only allows for 28 characters (rather
than 32).
48228: SoundPoint IP 320, 321, 330, 331, 335: Contact Directory has a
nonfunctional "<New Entry>" option and incorrect Navigation Cluster Guide
NCG while dialing.
48257: VVX 1500: Default background image is not displayed after the
following sequence of events: select an image file, followed by selecting an
invalid image (file not found) and then selecting the default background image.
48463: VVX 1500: Cannot view JPEG images with file extensions .jpe or .jfif.
48701: VVX 1500: The touch-screen becomes disabled during keypad
diagnostics.
48776: VVX 1500: Scrolling in the Ethernet menu may cause the selected
highlighted item to be positioned at the bottom of the screen.
48840: VVX 1500: Pressing the "Slower" and "Faster" soft keys cause the
update cursor to advance immediately.
49331: VVX 1500: Audio is lost when disabling the hands-free mode while on a
speakerphone call followed by placing the call on hold and then resume it.
50812: Changes to configuration options are lost without warning if you exit
from the Settings menu without passing through confirmation dialog.
50855: SoundPoint IP 320, 321, 330, 331, 335: An error message is not shown
when a contact is saved with an empty contact number.
51152: VVX 1500: Back arrow is not working as back-space when in the
"Display and Touch Screen Diagnostics" or "Media Statistics" screens.
51237: SoundPoint IP 320, 321, 330, 331, 335: In the “Server Menu”, the
“Server Password” option accepts digits instead of characters as default.
51656: Interactive MicroBrowser should timeout if "mb.main.idleTimeout>600"
51664: VVX 1500: Phone enters LCD Power-down mode in 3 to 4 minutes
instead of the time set by
“powerSaving.userDetectionSensitivity.officeHours=0".
51669: VVX 1500: After both SIP and H.323 Call Server parameters in Admin
Settings are reconfigured, only one dialog method should be offered to exit. A
reboot should not be required.
51947: VVX 1500: Cannot delete the URL by selecting it right-to-left and
pressing the backspace key.
51993: SoundPoint IP 320, 321, 330, 331, 335: Cancelling the deletion of a
contact appends an ellipsis to that contact's entry in the list.
52212: Phone will not restart while another extension on a shared line is in
use. The phone thinks it is active on a call preventing the request to restart.
52374: Options in the “Forwarding” menu are appended with an ellipsis after
returning from the selected option.
52438: SoundPoint IP 320, 321, 330, 331, 335: Typing in a fully filled field does
not prevent the cursor from advancing and overwriting existing content.
52447: VVX 1500: After placing 21 encrypted calls on hold, the phone locks-up
and reboots at the 22nd multiple encrypted call.
52590
: SoundStation IP 7000 and HDX Integration: The “Add Video” soft key
should not be accessible when flashing the POTS line to make a second POTS
call. While playing dial tone for second POTS call, pressing the “Add Video”
soft key and dialing a video number may cause the HDX to lock-up and reboot.
52629
: Phones only accept incorrectly formatted tel: URIs. The micro-browser
requires that all tel: URIs be of the form "tel://number". However, the '/'
character is not valid according to RFC 2806 sec 2.2. For backwards
compatibility, it should continue to accept (and ignore) any '/' character(s), but
the phone should also accept valid URIs without the "//".
52655: Upon disabling the "directory", saving a contact from the corporate
directory to the directory file will cause the saved contact to reuse the speed
dial index starting from 1.
52690: SoundStation IP 7000: The "Add Phone” soft key should not appear
while the “Call Type” is set to Conference-SIP and the phone is rebooted
55457: VVX 1500: When the dual protocol line is registered only to the
gatekeeper and not to the SIP server, this causes: (1) Hot dialed SIP URL call
is made via H323. (2) Dialog options do not appear when a hot dial URL call is
attempted.
55477: SRTP Key renewal does not occur during local conference calls.
55478: DHCP VLAN Discovery (DVD) is reported as not active when it actually
is.
55485: VVX 1500: The Camera Settings "Save" soft key loses its context-
sensitivity upon second visit to the menu option.
55514: VVX 1500: Calling into a Video Server causes the phone to connect the
audio portion of call but does not establish a video connection.
55560: VVX 1500: On occasion, the phone displays an incorrect call duration
timer value: e.g. 24:46:38 instead of 00:46:38. This happens while on an H.323
call to an RMX-2000.55641: VVX 1500: The Y-axis auto-scaling of the „Network
Load‟ graph is inaccurate.
55697
: Phone should not reject call with 486 if NOTIFY:Alerting is received
before the INVITE and reg.x.lineKeys and reg.x.callsPerLineKey is set to 1
55644: VVX 1500: Dialing an LDAP contact from the on-hook state via
termination uses the incorrect routing protocol.
55907: VVX 1500: Typing a "." or "#" causes the on-screen keyboard to
unexpectedly close and discard any edits.
55911: SoundPoint IP 320, 321, 330, 331, 335: Changing the text entry mode
causes the backspace soft key to disappear.
55929: SoundPoint IP 320, 321, 330, 331, 335: Even though the Navigational
Cluster Guide NCG indicates that the down arrow should not be functional,
pressing the down arrow did affect a change on the field by causing its font to
change.
55964: VVX 1500: The phone does not seize the only unregistered share line
using the „New Call‟ soft key, speaker and headset function key.
56046: The default value of the „Sound Effect Destination‟ parameter setting is
not removed from the override file when a new value is selected from the
menu option.
56057: Phones are de-registered upon receiving a large number of NOTIFY
messages for watch buddy enabled contacts.
56147: SoundPoint IP 550, 670: Adding „Contacts‟ that are longer than 10
characters or numbers are not truncated on the idle screen.
56156: VVX 1500: The "abc/ASCII" string remains in the title bar even after
leaving edit mode for a menu item.
56161: Emergency numbers matched against
„dialplan.routing.emergency.x.value‟ are not sent to servers listed in
57369: SoundStation IP 6000, 7000: The Contact entry in the „Local Contact
Directory‟ takes a long time to display.
57398: Phone displays "Please enter a contact" pop up message even after
adding a contact in the local contact directory.
57443: SoundStation IP 7000: The display flickers while making an outgoing
call.
57597: VVX 1500: Using the phone with an HDX, the phone does not transmit
video upon resuming a SIP call.
57615: VVX 1500: The „autohide‟ feature stops functioning when "PIN" is
pressed while the tool bar is sliding down out of view.
57849: Phone does not acquire the correct VLAN using LLDP on occasion
from a bootup.
57863: Phone does not accept a DHCP END (0xFF) option in a DHCP INFORM
response.
57958: In the „fail-over‟ feature, while re-registering, there is 32 second delay
before sending INVITE to the third server.
58023: VVX 1500: An call into a 3COM VCX audio conference server will cause
the phone to reboot.
58172: SoundStation IP 5000, 6000: „Hot-dial‟ numbers disappear from the
screen if there is an incoming call during the outgoing „hot-dialing‟ state.
58177: Blind transfer: in certain scenarios, when two phones receive a PSTN
call and two people attempt to blind transfer to an internal extension, they will
hear a series of “beeps” after pressing the "Send" soft key indicating that the
transfer was not successful. If they cancel the transfer and try again, the
transfer will complete properly.
58197: After upgrading from 3.0.0 to 3.1.3 RevC, you may notice a delay in the
audio signal when answering a call using the speakerphone.
58296: VVX 1500: H.323 digit-map routing files when the „reg.1.lineKeys‟
configuration parameter has a value of greater than 1 and „reg.1‟ is assigned a
SIP number.
58362: VVX 1500: Initiating a URL hot-dial call by pressing the '#' or '*' key
causes the “Enter URL” dialog to pop-up with the „#‟ character already
inserted into the field, even through the „#‟ character is not a valid SIP URL
character.
58464: A „Contact‟ cannot be saved from a „Corporate Directory‟ search result
into a „local directory‟. This is as a result of not checking the correct attribute,
i.e. SIP vs H.323.
58498: Within the „Re-registration on fail-over‟ feature, „Subscribe‟ does not
trigger the fail-over. The phone does not send the register request to the
second server after received an ICMP from the primary server.
58509: Within the „Re-registration on failover‟ feature, the phone sends an
extra „Register‟ request to primary server after the first fail-over.
58520: VVX 1500: Uni-directional Video Streaming interoperation issue with
Siemens Video Desktop Client ODC.
58574: SoundPoint IP 650: Re-registration on failover: Phone does not
invalidate an existing registration when it is registered with a Broadsoft
server.
58619: Line Authentication: line becomes unregistered when an invalid name
and password is entered from the menu options on the phone. The line
becomes unregistered until the phone is rebooted.
58782: The phone will set the Call Control 802.1Q Priority incorrectly when
using TCP. The value is set correctly when using UDP.
58785: VVX 1500: Phone does not append the MAC address to HTTP user
agent headers when configured to do so. Introduced in SIP 3.2.2.
58787: VVX 1500: Phone reboots immediately after making a call to an RMX
when the Camera Target Frame Rate is set to minimum.
58874: When using TCP preferred transport, the phone will not resend a 200
OK message after answering a call without receiving an ACK.
58906: Phone does not clear its BLA state table when receiving a NOTIFY
message with state = full after a SUBSCRIBE message.
58907: VVX 1500: The phone fails to send an INVITE SIP packet when the
configuration parameter msg.mwi.1.callBack="voicemail" and user presses
the „Messages‟ key.
58908: VVX 1500: With BootROM 4.2.1.0334, the phone sends a truncated
Option 60 message.
58913: The phone reboots when pressing the Messages key while “Message
Waiting Indicator” is disabled. When the phone has more than one registration and „msg.mwi.1.callBackMode=”disabled” and
„msg.mwi.2.callBackMode=”disabled”‟, the phone will freeze when the
“Messages” key is pressed. The phone will no longer respond to any further
key presses.
59129: The “Centralized Conference” feature fails when a URI is incorrectly
assigned to voIpProt.SIP.conference.address.
59262: A conference notification will cause the phone to lock-up and then
reboot. This is as a result of invoking the “BargeIn” feature on a specific
Asterisk implementation
59308: A retransmitted INVITE message results in a “400” response. This is in
violation of RFC 3261 section 17.2.1.
59430: VVX 1500: Call received from a mobile will cause the phone to display:
parameter „tcpIpApp.sntp.daylightSavings.enable‟ is set to „disabled‟.
: VVX 1500: Phone displays incorrect time after the configuration
59737: SoundPoint IP 320, 321, 330, 331, 335: The “Line Label” is not
displayed on the top line of the screen when using the HTML idle display
micro-browser page.
59777: SoundPoint IP 320, 321, 330, 331, 335: When using "NN#" speed dial
feature, the title displays “Directory” instead of “Speed Dial”.
59949: SoundPoint IP 320, 321, 330, 331, 335: Idle bitmap graphic is displayed
on the bottom of the screen. Only half of the display is utilized when
“ind.idleDisplay.mode=2” or „3‟.
59954: The phone will lock-up and reboot when a “Re-INVITE” message within
same dialog is sent to the phone immediately after sending a “CANCEL”
message for the initial “INVITE”.
59967: VVX 1500: LDAP: When an incorrect CA certificate is installed, the
phone will not attempt to retry a TLS handshake.
60013: SoundPoint IP 320, 321, 330, 331, 335: The phone will lock-up and
reboot when accessing the contact directory if “dir.local.readonly=1”.
60126: Gateways reject an “INVITE” message when “reg.1.csta=1”. The
“INVITE” should include the header: “Accept:
application/sdp/application/csta+xml”.
60145: SoundPoint IP 650: The phone incorrectly presents 2 BLA call
appearances when only 1 should be shown. The 2nd call appearance
incorrectly indicates a remotely held line, when it is not.
60264: SoundPoint IP 450, 550, 560, 650, 670: When a BLA line is showing the
dialing screen, remote call appearances should not be displayed when the
remote BLA line resumes a call.
60266: SoundPoint IP 320, 321, 330, 331, 335: When a phone is in dialing
screen, if a remote SCA/BLA line holds and resumes, the dialing icon is
changed between animation arrow and termination (speaker) icon. The
termination icon should be displayed continuously and should not change.
60267: SoundPoint IP 550, 560, 650, 670: Cannot change a checked item twice
in the "Prioritize Background" menu.
60340: SoundPoint IP 650: The “Join” soft key should not be displayed on a
phone with a BLA line when there is only one call on the phone.
60650: VVX 1500: The idle browser will alternate between current content and
earlier content when it the display is refreshed.
62621: SoundPoint IP 321, 331: Phones running SIP 3.2.3.3122 and configured
for HTTPS are displaying error messages: “Alert:Fatal, Description: Decode
66743: Phones may be vulnerable to Denial of Service attacks when used in
certain configurations. Sending HTTP GET requests with a broken
authorization header can produce a device restart under certain
circumstances in certain models of phones. For full details, refer to Technical
Bulletin TB66743. See Section 4 Reference Documents for the location of the
documents.
2.5 Version 3.2.4
2.5.1 Added or Changed Features
N/A
2.5.2 Removed Features
N/A
2.5.3 Corrections
59308: A retransmitted INVITE message causes a “400 Bad Response” reply.
This is in violation of RFC 3261 section 17.2.1.
65207: A consistent but slow memory leak occurs as a result of receiving
INVITE messages containing “replaces”.
65435/65725: SoundPoint IP/VVX 1500: [IEC 60268-1]: The default and
maximum values for the headset and headphone audio levels have been
adjusted to ensure compliance with the IEC 60268-1 TUV safety requirements.
65660: The BootBlock may become corrupted as a result of accessing
43099: Added support for SoundStation IP5000 Conference Phone.
43297: Sound effects can now be played out of a destination based on user
configuration. Configuration parameters: se.destination= “chassis”,
“handset”,”headset” or “active”. Default is “chassis”.
45462: All SoundPoint and SoundStation phones now comply with “retry-
after” instructions embedded in SIP Response codes 500 and 503 as part of
REGISTER and other requests.
50739: SoundStation IP7000 – HDX Integration: On a multi-leg conference,
when the 'End Call' soft key or the 'On Hook' hard key is pressed, the
conference phone will ask the user if the entire call should terminated. A
negative response will guide the user to the conference “manage” menu to
allow the user to terminate the individual legs of the call. The dialog only
appears for multi-leg conference calls.
51753: SoundPoint IP 450: Improved the appearance of anti-aliased characters.
51940: All SIP phones now have a “fail-over” feature that enables phones to
re-register before diverting SIP signaling to an alternate server.
NOTE: This feature will be formally released and documented in a future
release.
54041: Format of DHCP Option 60 Data is now configurable and added support
for Option 125 as per RFC 3925.
54983: VVX 1500: Internal IP address of phone (instead of an alias) is no longer
being sent in the Facility Message.
55524: Logs no longer display "Can't set 802.1Q VLAN id for TCP protocol"
messages at default when running on a VLAN.
56272: Network Configuration DHCP sub-menu now supports Option 60
format. The new options include setting either “RFC 3925 Binary [default]” or “ASCII String”.
2.6.2 Removed Features
None.
2.6.3 Corrections
45188: SoundPoint IP 320, 330, 430: The minimum acceptable amount of free
RAM has been increased in order that functions such as ring-tones are not
affected.
47897: „Back‟ soft key is not working when user tries to exit from Instant
Message menu.
52119: VVX 1500: Phones may reboot during G.729 packet loss concealment
52787: voIpProt.SIP.requestValidation.x.method="source" does not work with
DNS SRV Static Cache
53473: SoundStation IP 7000: When used with an HDX, the parameter
voice.volume.persist.handsfree ="0" has no effect on the HDX.
54549: SoundPoint IP 450: Changes in the display color palette have created
contrast problems.
54751: SIP Invite Message is not sent when dialing a number containing the
period character. When a call is placed using a following number with a
period, e.g. "12.345.6789", the INVITE message is not sent to "12.345.6789".
The phone misinterprets the number as an IP address and attempts a DNS
lookup for '12.345.6789' without success.
54832: VVX 1500, IP 321, 325, 330, 331, 335: Phone allows user to add more
than 32 characters in Hot Dial screen.
54867: SoundPoint IP 321, 325, 330, 331, 335: In the Contact Directory, the text
fields do not scroll to the left to reveal the first character until you actually
move the cursor to the first character.
54908: SoundPoint IP 321, 325, 330, 331, 335: A „colon‟ „:‟ is unexpectedly
displayed in the scrolling status line during an incoming call.
55099: VVX 1500: Steering video between "active" and "inactive", the video leg
fails in a long SRTP conference.
55120: SoundPoint IP 550, 560, 650, 670: Dialing numbers in “Contact
Directory” unexpectedly opens contacts for editing.
55296: VVX 1500: The dialpad widget is not presented when attempting to
conference or transfer a held call while in a ringback state.
55378: VVX 1500: Phone fails to invoke LCD power down mode after remote
end places the call on hold.
55415: Phone allows the user to enter more characters than it is capable of
saving in the Contact Directory fields. Introduced in SIP 3.2.0.
55420: VVX 1500: Phone fails to play back video after a SIP re-INVITE message
is sent to RMX meeting room.
55560: VVX 1500: Phone displays incorrect call timer values while in an H.323
call to an RMX-2000.
55618: SoundPoint IP 450, 550, 560, 650, 670, 5000, 7000: Switching to
Katakana characters before the character selection widget times out,
produces random characters that on occasion causes the phone to
malfunction.
55844: SoundPoint IP 321, 325, 330, 331, 335: Proceeding outgoing call state
on one line is adversely affected by an outgoing call on another line.
55884: SoundPoint IP 650: On occasion, the display freezes and both BLF
Extension Modules‟ display may become blank during a consultative transfer.
The phone does not recover and has to be rebooted.
56032: SoundPoint IP 650 + 2 Expansion Modules: On occasion, the phone will
reboot while monitoring continuous BLF traffic.
56488: SoundStation IP 6000, 7000: DHCP client asks for duplicate options. In
packets sent from the client, the "Parameter Request List" option contains two
requests for the options "Router"(3) and "Domain Name"(15).
56641: SoundStation IP 6000, 7000: Intermittently ignores the LLDP broadcast
from a switch. It will default to the data VLAN instead of the voice VLAN. There
is a LOSS of LINK during the boot process causing LLDP to fail.
56836: SoundPoint IP 550, 560, 650, 670: Lifting the handset unexpectedly
dials the last hot-dialled number immediately after adjusting the volume.
57133: SoundPoint IP 321, 330, 331: Phone does not display a customer
supplied logo. It is displayed for only a fraction of a second after a reboot.
57457: LoudRing.wav audio file is not distributed in release 3.2.2.
57796: Invalid Message-Summary Event results in invalid MWI notification.
57849: SoundPoint IP 330, 550: Phone is not acquiring the correct VLAN via
LLDP. The phone is "losing link" somewhere during its boot process. When
this happens, the LLDP neighbor ship will be torn down and this in turn forces
the phone to default to the wrong VLAN.
58024: VVX 1500D: Hold function fails in a specific customer scenario.
41450: VVX 1500: Change of the real time operating system.
43760: VVX 1500:H.323 signaling protocol support for video.
43862: VVX 1500: Add support for Webkit browser to replace the XHTML
browser.
45172: VVX 1500:Add support for iLBC audio codec.
47173: VVX 1500: Add support for H.261 video codec
48557: VVX 1500: SetDefault max video bit rate to 384 kbps
48743: VVX 1500: Upgrade curl library to version 7.19.
48961: VVX 1500: Add support for H.235 security
49069: SoundStation IP 6000, 7000: Add support for iLBC audio codec
49079: VVX 1500:Add support for mutual TLS authentication.
49277: VVX 1500: Add support for LLDP protocol.
49430: VVX 1500: Add ITU-T G.719 vocoder
50125: VVX 1500: Outgoing calls support dual (SIP/H.323) protocols
51084: VVX 1500: Add support for video fast update request via RTCP, RFC
5104
52944: VVX 1500:Add menu support applicable to H.323 usage.
53849: Formalize support for “DTMF via SIP INFO” (initially supported in SIP
3.2.0)
54025: Increase maximum size of contact directory to 128 to facilitate complex
dialing scenarios.
54239: VVX 1500: Add user accessible menu option to select the video call
rate. Default configured using video.callRate.
2.7.2 Removed Features
52522: VVX 1500: Remove “Launchpad” Feature.
2.7.3 Corrections
44782: VVX 1500: Improve phone UI response when a local conference is
active.
44980: VVX 1500: Fall back to configured video codec configuration for Tx
video when incoming signalling lacks codec modifiers
47023: VVX 1500: Occasionally the text font changes.
47476: XML API: When the user is inside an XHTML Form Field the Submit
soft-key does not show up
47768: SoundPoint IP 450: CDP power usage advertisement is low for peak
power conditions.
48175: VVX 1500: Conference not established using EFK feature.
48784: VVX-1500: Softkeys not restored after rejecting a call from within the
„Applications‟ UI context.
48857: VVX 1500: Recording (R) stops or reboots phone in various high load
scenarios such as (a) recording during SRTP conference call, or (b) recording
while browsing the application menu during non-SRTP conference call
48921: VVX 1500 Digit key presses may be missed in certain scenarios
50152: VVX 1500; Corporate Directory: Change non-null sticky primary filter,
search (filtered) bar remains on old data
50192: VVX 1500: Media Statistics menu is not displayed correctly for several
languages
50286: VVX 1500; Corporate Directory: Pressing page down key "#" does not
move entry list after pressing page up key "*" in quick search menu
50531: SoundStation IP7000: Phone will not startup without network
connection when using the PIC cable
50624: Inbound call is rejected due to timeout but no 603 is ever sent because
TCP stream has already been reset.
51141: Remove the small number on the left side of the scrolling status bar
51449: VVX 1500: Out of Dialog Refer based dialing is failing. SDP on INVITE
from VVX is missing media attributes, generating a 606 response.
51533: Backlight intensity change is not updated appropriately in Overrides
config file.
51605: VVX 1500: Push request will get lost if it follows another push request
immediately.
51643: SoundStation IP 6000, VVX 1500: Japanese Language is not properly
displayed.
51753: SoundPoint IP 450. Display text look fuzzy especially when using Asian
fonts
51959: Handling of Hold re-Invites is incorrect after one-touch blind transfer to
full park orbit.
51965: HTTP request messages are not directed to proxy
52164: VVX 1500: Hot-dial does not work in headset mode.
52360: 'Auth Password' field' can be viewed in web configuration page.
52365: Phones don't transition very well from LLDP to CDP.
See Administrator‟s Guide
for SIP 3.2.2 for details
sip
added
ind.anim.IP_335.44.frame.2.bitmap
PlumHd
See Administrator‟s Guide
for SIP 3.2.2 for details
sip
added
ind.anim.IP_335.44.frame.2.duration
1300
See Administrator‟s Guide
for SIP 3.2.2 for details
2.9 Version 3.2.1
2.9.1 Added or Changed Features
None.
2.9.2 Removed Features
None.
2.9.3 Corrections
53322: Setting voIpProt.local.port to a non standard port does not send from
or advertise that port
53611: User Language Selection is lost on Upgrade to SIP 3.2.0
Note that the fix for this issue will guarantee retention of language setting
when upgrading from releases prior to 3.2.0 (e.g. 3.1.3) but WILL NOT preserve
language changes made when the phone was running SIP 3.2.0 .
53685: Phones ignoring nat.ip parameters.
53852: SoundStation IP 7000/HDX Integration: DTMF duration should be set to
26754: SoundPoint IP 320,321,330,331,450, 550, 560, 650, 670: Add support for
the iLBC codec
30079: Add support for mutual TLS authentication. See technical bulletin
TB52609 and the Administrator‟s Guide for more details on this feature.
32259: Recognize multiple mime types in the microbrowser.
32753: Add support for LLDP protocol. To take full advantage of this feature
BootROM 4.2.0 should be used.
34782: Replace libSRTP algorithms with OpenSSL versions
35525: Modify DND Status Message.
37118: Add ability to invoke a „screen capture‟
39358: Add a „Loud Ringer‟ Ring-Tone selection. See technical Bulletin 39358
for instructions on how this can be configured.
30855: SoundStation IP 7000: Create a SoundStation IP 7000 Setup Guide.
41579: Meet requirements of ETSI TS 102 027-2 v4.1.1 RFC 3261 compliance
test for Anatel/Brazil
43141: Add support for „Statically Configured‟ BLF and Call park and retrieve
enhancements
43142: Add support for single button Blind Transfer and Retrieve of a call
designated as an „automata‟ in the Dialog used for „Statically Configured‟ BLF.
43646: Improve boot-speed in some situations where the boot server is
incorrectly configured.
45057: Languages selection presented in appropriate language
45174: Upgrade zlib to version 1.2.3
45743: Upgrade curl library to version 7.19.2
45787: SoundPoint IP 450, 550, 560, 650, 670: Add instructions for changing
label colors in the User Guides.
45791: SoundStation IP 7000/HDX Integration: Add a configuration option to
disable Digit-map rules for „Remote Dialing‟ when connected to an HDX.
46093: Add ability for User to enable/disable display of idle browser from
menu
46113: SoundPoint IP 320, 321, 330, 331: Add navigation button „shortcuts‟ in
46248: SoundStation IP 7000/HDX Integration: Add Admin menu option to
manually specify the value to be used as the „extension‟ displayed on the
phone screen.
46424: Improve readability of Menu items when using Background images on
the display.
46446: Provide a menu option to view the status of feature licenses.
46683: Remove Background from scrolling Status Bar for improved
readability.
47355: Scrolling Status Bar should give equal time to each status message
47390: Add configuration parameters for select ETSI SIP compliance
requirements
47463: Allow for secure entry of passwords in the micro-browser API
47487: Forcing a 'Back' soft-key in the micro-browser soft-keys is
cumbersome
47689: Add support for SoundStation IP 7000/HDX6000 Integration. This
feature requires a future update release to the HDX6000 software.
47749: Support Transmission of Join Header as per RFC 3911
48004: Add support for BLF call pick-up using Dialog-info within an INVITE
with Replaces header
48055: Enhanced BLF: Improve user experience when an incoming call occurs
whilst the user is viewing BLF monitored line call details.
48109: Include "fmtp" attribute specifying Mode=30 in the SDP when 13.33
kbps iLBC is used.
48136: Remove platform specific TFTP code and instead use tftp support in
curl library 7.19.2
48137: Add support for BLF call pick-up using Dialog-info within an INVITE
with Replaces header
48205: SoundStation IP 6000, 7000: Add support for the iLBC Codec.
48559: Scrolling status line should have similar look on various phones.
48578: SoundPoint IP 430: Reduce the local Contact Directory maximum to 99.
48579: SoundPoint IP 430: Reduce the maximum number of calls supported to
4 (from 8).
48664: Add User accessible menu option to display whether a device
certificate is installed.
48678: During local conferencing it is now call diagnostics for each call leg.
Accessed from Menu->Status->Diagnostics->Media Statistics.
48738: Add configurable behavior for Directed Call Pick-Up as used for
48780: Add option to apply digit-map rules to tel:URI initiated calls
48846: Add configuration option for whether the call appearance on a remotely
monitored BLF line should be presented on the monitoring/attendant phone.
48861: Add configuration option voIpProt.SIP.strictReplacesHeader to control
whether the phone requires call-id,to-tag and from-tag to perform and INVITE
with Replaces.
48984: Phone will populate the display-name field in the To header of
responses that it generates
48998: Add configuration option for the phone to send 486 Busy when call is
rejected.
49309: Combine SoundPoint IP 550 and 560 User Guides.
49465: Update Destination of outbound call based on display-name in SIP To
header responses
49660: Call Forward: "user=phone" should be included in "refer-to" parameter
of Refer: header
49695: Allow for SDP offer or answer in provisional reliable response and
PRACK request and response
49839: RTP Rx must detect and correct for G.722, G.722.1, G.722.1C, and G.719
RTP timestamp increments based on different sample rates
50769: SoundStation IP 7000/HDX Integration: Add support for Hook-Flash
during POTS calls.
50927: Add Equifax Secure eBusiness CA-1 to the trusted CA list.
51419: RFC2543 hold not working when video SDP present in certain
scenarios
2.10.2 Removed Features
48283: Remove support for SoundPoint IP 301, 501, 600, 601 phones.
48698: Remove support for SoundStation IP 4000
2.10.3 Corrections
27048: Application load progress bar doesn't match actual progress
29148: Phone doesn't format the file system when it notes error on screen
while loading large configuration files.
29344: HTTP Digest Authentication does not work on IIS.
30219: Logs are not uploaded when phone resets to factory default
31858: Shared line indicator led turns off when 2 phones resume
simultaneously
34681: stickyAutoLineSeize and call.enableOnNotRegistered="0" seize wrong
35288: Config web-site takes too much memory during initialization
35991: Roaming Buddy list with Office Communicator reports all buddies as
offline
36969: SoundStation IP 6000 doesn't display Japanese language correctly
38348: SoundPoint IP 320, 321, 330, 331: SRTP call displays incorrect line
icons in a certain scenario.
38392: Blind Transfer from encrypted phone to an unencrypted private line
does not establish the new call as encrypted
38418: Phones sometimes show SRTCP authentication failure at log level 0
38824: After audio diagnostics (i.e Record and Play in handset), 1st call gets
established in handset mode even if the handset is ON-HOOK.
39013: SoundStation IP 7000 should not recognize cell phone cable without
physical cell phone attached
39143: P-Asserted-Identity header in initial INVITE message not used for caller
40679: SoundStation IP 6000: Changing the status on "MyStatus" menu does
not change the OC client status when roaming_buddies.reg = 1.
40892: SoundStation IP 7000: There is no Time/Date displayed as first phone
call established.
41939: Call Recording: User is not able to play the wav file when it has a "call
on hold" and also in "remote busy state". Junk characters appear in audio
player.
42092: Special Slovenian characters not included in phone's fonts
42213: SoundStation IP 7000: There is no "SIP:" string displayed whe using
URL dialing.
42611: USB Call Recording: When full USB drive is attached recording should
not begin and no new file should be created
42761: SoundStation IP 7000/HDX Integration: Pressing Content soft key on
SoundStation IP 7000 prompts the user to choose VGA input
43910: Microbrowser fails to process http response with image/bmp directly in
a certain situation.
43916: Some of the configured sampled wave files are not downloaded onto
phone becuase of insufficient RAM Disk size.
43990: SoundStation IP 7000: Missing glyphs in the Katakana bitstream fonts.
44100: If aCall display name includes an @ then the display is truncated after
44248: Micro Browser not displaying any error message when an unsupported
media configured in the microbrowser URL.
44273: When SIP Contact header is a comma separated list only the first
contact is processed
44278: Phone number is not displayed correctly on line key when the length of
phone number is more than 10 characters.
44301: SoundStation IP 6000,7000: Date is not displayed when idle browser is
enabled
44377: Redial key cannot be reassigned
44443: SoundPoint IP 320,321,330,331: Menu exit via Menu key is not ignored
while in Edit mode.
44635: SoundStation IP 6000: Phone uses incorrect configuration parameters
to download customizable fonts
44783: Cipher list displays different items for different TLS transactions
44844: USB Call Recording: Stopping Playback through "Back" key not
intuitive
44855: Call Lists: Missed Calls not incremented on Call Forward on Busy
44892: SoundStation IP 6000, 7000; SCA Barge-In: Phone barges in to the
wrong call in a certain scenario.
44962: Phone displays 3-way animation icon in held screen when conference
legs on hold
45143: Centralized Conference: When max conference size is reached phone
displays local conference UI
45327: Establish a call between two phones configured as shared lines, press
down arrow key, all soft keys disappear
45428: Unexpected re-INVITE occurs before BYE, when removing a leg from a
conference call
45650: Double hold w/ MOH and a non-Polycom SIP phone: one way audio -
MOH fails
45658: Platform string in transmitted CDP packets is not consistent across
SoundPoint IP products.
45716: SoundPoint IP 450: Text is not as clear as on other phones.
45835: SoundPoint IP 450: Status Bar text is difficult to read on some
backgrounds.
45943: Incorrect logic used when picking line for outgoing call in a multiple
registration scenario.
46068: “Transfer On Proceeding” is not supported when server is a proxy
46334: DTMF local rendering does not stop if far end holds while local digit
46478: EFK: Phone does not send invite when executing $Cwaitdialtone$
46513: Dialog Event Package Content Guideline 6B (Local Identity)
46514: Dialog Event Package Content Guideline 6C (Local Target)
46547: SoundStation IP 7000: Warning Header Text notification does not
display on phone (when configured)
46550: Directed-Call-Pickup fails when SIP server is a proxy.
46588: SoundStation IP 7000/HDX Integration: Info Soft key is missing in
Contact Directory
46738: Enhanced BLF: attendant.ringType parameter is not removed from the
override file when default (silent) attendant ring type is selected
46741: Enhanced BLF: The remote call appearance screen does not time out
on console phone until the watched line hangs up an outgoing call
46770: Microbrowser: * and # buttons do not work correctly when text input
mode is set to numeric on input fields
46899: Electronic hook switch: No audio during active call if answer by
pressing hook switch button immediately on Jabra headset under specific
scenario.
47039: The line LED does not flash instead remains stable green, when an
active call is kept on hold during an incoming call.
47123: USB Call Recording: Missed call notification is getting displayed on the
audio player screen if an incoming call is not answered during playback
47207: SoundStation IP 7000/HDX Integration: When the MUTE is active it
covers up the dialing fields so I cannot see what I am dialing
47248: Hot dial doesn't work when lifting the handset for the second call when
call.stickyAutoLineSeize="1"
47300: URL dial disabled message never displayed - Failed to route to
voicemail from "Message Center" tab
47336: SoundStation IP 7000/HDX Integration: Received\Missed call list is
showing IP address of SIP server instead of the Extension number of a call
received/Missed from a SIP extension.
47464: SoundPoint IP 320/330; SoundStation IP 7000: When two incoming calls
are active on a phone lifting the handset or pressing the hands free key to
answer the call results in the most recent call being answered even though
the ring-tone is played according to the first incoming call.
47535: Soft keys reset to default layout on an inbound call in some multiple
call handling scenarios
47566: XML API; Internal URIs: When a internal URI is executed with multiple
VolUp and VolDown action uri's, the Ringer horizontal bar is not seen, only the
Volume sound going UP and Down is heard.
47951: Transfer should have precedence over pickup of a ringing BLF line
when pressing the linekey during a call transfer
47953: SoundStation IP 6000: Call info display not displayed properly when
volume up/down key press.
47958: SoundStation IP 7000/HDX Integration: Unable to add more than one
contact dir when Onyx is configured with no Ethernet cable connected + HDX
47962: SoundStation IP 7000/HDX Integration: Incorrect icon displayed when
Redialing POTS call but there is nothing in the buffer to redial. Phone should
not attempt to dial when redial buffer is empty for the call type selected.
48003: SoundStation IP 7000/HDX Integration: Phone dials POTS call as video
call when dialing from idle state for a certain configuration.
48011: SoundStation IP 7000/HDX Integration: Use of the Idle Browser
interferes with some display elements e.g. Mute Icon, Video/Phone Call Pop-up
when connected to HDX.
48019: SoundStation IP 7000/HDX Integration: The pop-up message "Video or
Phone Call?" is overwritten by idle browser
48045: Enhanced BLF: Phone does not hold the 1st call when press Dial soft
key to make the 2nd call to the same called party
48049: BLF: Attendant phone does not display all remote calls on a BLF
monitored line if the Monitored Phone has a call in the „Ringing‟ state.
48061: Enhanced BLF: Attendant phone does not update 1/x widget when BLF
monitored line has 1 or multiple incoming calls being ended
48069: U/I : SCA Barge-In: Extra softkeys are displayed on remote shared
phone while viewing call appearance list by long pressing line key
48071: XML Push API: Key:Handsfree internal URI action is not executed by
phone in a certain scenario.
48115: SoundStation IP 7000/HDX Integration: HDX plays ring sound after
answering POTS call
48131: Call Forwarding Status Not Always Shown if multiple Call Forward
Types are selected.
48149: SDP attribute truncated when first character of the value is a digit
48162: "Boot Server" status field shows incomplete or blank path if a “/” is
included in the setting.
48174: Failed call may cause subsequent calls to skip URL/Number mode
selection
48179: XML API; Telephony Notifications: Called Party number is shown
overlapped in incoming event notification in case of IP dialed calls between
unregistered phones.
48209: Cannot delete left-most character before character selection timeout
48981: SRTP fails in 3.1.2 when the user presses Hold then Resume during a
call. This happens on several different models of IP phone.
48996: Phone not tagging correct DSCP value to some packets (Trying,
Ringing and OK)
49106: Entire dialed URL is not always saved in call history
49251: Update Polish XML Dictionary to include Polish characters
49300: SoundStation IP 7000/HDX Integration: Insure that DTMF tone are being
sent via the dtmf start/stop Clink2 API
49417: Phone reports MOH dialog if SUBSCRIBE received while on hold
49459: Cancel doesn't work after entering hotdial digits.
49461: DND symbol(X) does not disappears after DND feature is disabled in a
certain configuration.
49473: SoundPoint IP 320,321,330,331;Corporate Directory: If I use the # key to
change text entry mode it should reset the Quick Search timeout timer
49476: Corporate Directory: Scrolling indicators work poorly
49512: XML: HTTP Refresh header response is not loading the specified URL
on the phones after the specified amount of time has passed, in a certain
situation.
49516: Hanging up handset does not terminate call in Audio or Display
Diagnostics
49523: SoundPoint IP 450, SoundStation IP 7000: Asian fonts appear „fuzzy‟
49548: SoundPoint IP 320, 321, 330, 331: Edit and Delete softkeys remain after
49620: Volume settings for Recording do not work in handsfree mode.
49639: Handsfree dial tone is interrupted by hold reminder and call waiting
ring tones
49641: SoundStation IP 6000, 7000: Call info display does not display properly
while changing volume.
49677: Phone does not comply with rfc4475 3.1.2.3. Negative Content-Length
49685: SoundPoint IP 320, 321, 330, 331: Cannot enter URLs with uppercase
letters
49692: SoundPoint IP 450 : Seconds Colon in time does not blink for every
second.
49693: ACD icon not displayed when parameter
(voIpProt.SIP.serverFeatureControl.cf=1) is enabled.
49696: After a long LAN outage during "Downloading new application" the
phone is re-connected to the network. It gets back an IP but it does not reboot
and it does not display any error message
49701: SoundStation IP 7000/HDX Integration: Phone response with
"reg.1.server.1.expires = "5" setting is inconsistent
49706: SoundStation IP 7000/HDX Integration: SIP Extension display disabled
after dis-connecting from HDX with HDX-Preference option
49757: SoundStation IP 7000: Phone does not display "Network Link is Down"
after the cable is disconnected from a hub
49758: SoundStation IP7000: Phone gets into a bad state and does not recover
from temporarily unplugging network connection during an active call.
49776: If dir.corp.user is mis-configured, the phone does not display "Login
Error"
49813: Corporate Directory: Phone displays 'Enter More Chars...' when
submitting a string that returns no results in the Quick search mode.
49825: Corporate Directory: Black background for Search bar displays
inconsistently on different platforms
49829: NTP Time synchronization unreliable in a particular scenario.
49834: Corporate Directory: If VLV indexing is configured and an Advanced
Find yields more results than the configured „pageSize‟ (Default is 64)
scrolling through the entries may not work correctly.
49836: Corporate Directory: Phone flashes "Please try again" msg for 1 time if
Corp Dir server is down->phone reboots up->Open Corp Dir menu
49911: Incoming ring tone not played on the phone in a certain enhanced BLF
use case.
49926: SoundPoint IP 320,321,330,331: Phone auto-increments new contact's
speeddial index to 100 even though the maximum entries is 99.
A flag to determine if the
dial plan applies to
for calls made through the
Polycom HDX
system.
sip
added
dialplan.applyToTelUriDial
0 or 1
Default is 1
A flag to determine if the
dial plan applies to
uses of the tel:// URI.
sip
added
ind.class.2.state.35.index
44
Changes Relating to
screen layout
sip
added
ind.class.2.state.36.index
42
sip
added
ind.class.2.state.37.index
43
51631: Phone not releasing first assigned IP address when VLAN is set via
DHCP.
51633: Phone fails to play busy/reorder tone upon a refer based transfer when
it gets a 603 or 486 response
51644: Some Japanese strings do not display correctly.
51690: EFK feature is used for onetouch Voicemail dialling. When using on
3.1.3 the phone appears not to honour the stickyautolinesieze
51718: Phone continues to ring after the call has been answered with a certain
call signaling sequence.
51763: SoundStation IP 7000/HDX Integration: When Adding video to an
existing call. IP7000 shows as on Mute but Far end can hear them.
51838:Some Japanese characters are not properly displayed.
52014/53597: In SIP 3.x.x when an IP phone picks up a transferred call in a
certain scenario, the call is immediately placed on Hold instead of being
connected.
52017: Web interface issue Password entry is not masked when entered (since
SIP 3.0.0)
52108: Phone fails to restore destination to Asserted Identity or Remote ID
after a transfer fails
2.10.4 Configuration File Parameter Changes
This section lists the parameters that have been added/changed or deleted from the
template phone1.cfg and sip.cfg files. For further description of parameters please refer to
the Administrator‟s Guide for the SIP 3.2 Release.
Note also that the template 000000000000.cfg file has been modified in order to facilitate
support for the Legacy phones and the VVX 1500 in this release.
available to the user. The
user can then decide
whether or not the
background takes priority
over the idle browser.
Used in conjunction with
up.idleBrowser.enabled .
sip
added
up.screenCapture.enabled
0 or 1;
Default is 0
A flag to determine
whether or not the user
can get a screen capture
of the current screen
shown on a phone. The
flag is cleared when
the phone reboots.
sip
added
voice.audioProfile.iLBC.13_33kbps.
payloadSize
30
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
voice.audioProfile.iLBC.15_2kbps.p
ayloadSize
20
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
voice.audioProfile.iLBC.jitterBufferM
ax 160
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
voice.audioProfile.iLBC.jitterBufferM
in 40
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
voice.audioProfile.iLBC.jitterBufferS
hrink
500
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
voice.audioProfile.iLBC.payloadTyp
e 110
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
removed
voice.audioProfile.Lin16.44.1ksps.p
ayloadType
120
Parameter renamed.
sip
added
voice.audioProfile.Lin16.44_1ksps.p
ayloadType
120
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
voice.audioProfile.Lin16.8ksps.paylo
adType
116
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
voice.codecPref.iLBC.13_33kbps
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
voice.codecPref.iLBC.15_2kbps
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
voice.codecPref.IP_6000.iLBC.13_3
3kbps
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
voice.codecPref.IP_6000.iLBC.15_2
kbps
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
voice.codecPref.IP_650.iLBC.13_33
kbps
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
voice.codecPref.IP_650.iLBC.15_2k
bps
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
voice.codecPref.IP_7000.iLBC.13_3
3kbps
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
voice.codecPref.IP_7000.iLBC.15_2
kbps
See Administrator‟s Guide
for SIP 3.2.0 for details
If set to 1, an SDP offer or
answer is
generated in a provisional
reliable response
and PRACK request and
response.
If set to 0, an SDP offer or
answer is not
generated.
If set to 1 or Null,
validation of the SIP
header
URI scheme is enabled.
sip
added
voIpProt.SIP.strictReplacesHeader
This parameter applies
only to directed call
pick-up attempts initiated
against monitored
BLF resources.
sip
added
voIpProt.SIP.use486forReject
If set to1 and the phone is
indicating a ringing
inbound call appearance,
phone will transmit
a 486 response to the
received INVITE when
the Reject soft key is
pressed.
Remove support for the SoundPoint IP 320, 321, 330, 331, 430, 450, 550, 560,
650, 670 products.
Remove support for the SoundStation IP 6000, 7000 products
Remove support for the VVX 1500 product.
2.13.3 Corrections
50189: SIP responses missing to-tag after Phone challenges INVITE
51031: Cannot change the language to Russian
52237/52017: Web interface Password entry is not masked when entered
(since SIP 3.0.0).
53826/50546: When URL dialing disabled, BLIND soft key appears in the 4th
soft key slot, as opposed to the 3rd slot, after pressing TRANSFER.
53827/51690: EFK feature is used for onetouch Voicemail dialing. When using
SIP 3.1.3 the phone appears not to honour the stickyAutoLineSeize.
53828/52014: In SIP 3.x.x when an IP phone picks up a transferred call in a
certain scenario, the call is immediately placed on Hold instead of being
connected.
53829/50254: Phone does not honor SDP sent in PRACK.
54214/50869: Phone will only offer SRTP when SRTP crypto suite is selected
2.13.1 Configuration File Parameter Changes
None.
2.14 Version 3.1.3 C
2.14.1 Added or Changed Features
Add Support for the SoundPoint IP 321 and 331 products.
48854: Change default for parameter mb.main.idleTimeout from 20 to 40
seconds.
48567: When DND/CF Sync is enabled the phone should not Forward or deny
any calls that it receives
2.16.2 Removed Features
47376: Remove License Requirement on uaCSTA feature
2.16.3 Corrections
23634: SoundPoint IP 320/330, 430, 450, 550, 560, 650, 670, SoundStation IP
4000, VVX 1500: Packet stats jitter should be computed exactly as shown in
RFC3550. Issue remains on SoundPoint IP 301, 501, 600, 601 and
SoundStation IP 6000, 7000 phones.
43517: REFER-based 'click-to-dial' causes errors and may cause a phone
reboot.
44973 SoundPoint IP 301: Line label disappears after SCA phone views remote
shared line's call appearance list and the view screen times out
46795: SoundPoint IP 450: Colon in time display does not blink
46480: SoundPoint IP 301, 501, 600, 601:Loud static „pop‟ and „hiss‟ may be
heard when receiving audio using G.729AB as the codec with VAD enabled.
46613: SoundPoint IP 301, 501, 600, 601; SoundStation IP 4000: Audio not
transmitted or routed via default gateway when phone‟s subnet mask does not
match phone‟s IP address network class.
47303: URL BLF speed dial calls are using the incorrect "@domain" in
Signalling in certain scenarios.
47492: SoundPoint IP501: Message LED flashes continuously after receiving
blind transfer from a „centralized conference‟ leg
47609: SoundPoint IP 450: Phone is not able to display more than two status
notifications if server controlled ACD is enabled
47878: CLONE -Phone generating malformed XML with ACD Login/Logout for
some parameters.
47911: Forked INVITE back to caller fails to connect to voicemail on call
timeout
47915: Phone ignores 401 challenge after responding to 407 in a certain call
scenario.
47960: SoundStation IP 7000/HDX: Redialing POTS call from placed call list
dials as video call if the call was dialed from contact directory.
47964: SoundStation IP 7000/HDX: Phone displays wrong icon when
conferencing and adding a POTS call
48002: SoundStation IP 7000/HDX: Speaker volume drops to two bars after
48756: Unknown Party displayed on caller ID when using a shared line and
only number is provided, no name.
48778: VVX 1500: Motion detection is not starting after a video conference call.
48858: BLF attendants monitoring both initiator and recipient get confused
about state when initiator and recipient use the same dialog ID
48912: REFER transaction timeout set too high due to subscription state
expires from a NOTIFY with sipfrag on a successful blind transfer
48920: IP7000/HDX: When placing a Video conference call with 8 legs, the UI
does not show the two last call appearances.
48959: SoundPoint IP 430: After upgrading to SIP 3.1.2, the time portion of
date and time cut off when using a custom Idle Display.
48985: The phone may reboot if you receive or miss a call while looking at
information about a previously received or missed call.
49013: DND X icon does not update next to line key when BroadWorks ACD is
enabled.
49068: Receiving an OPTIONS message results in a spurious dialog
Notification being sent
49129: VVX 1500: U/I not showing updates while soft keys, physical buttons do
work.
49181: VVX 1500: When using the idle micro-browser the phone display
sometimes freezes‟.
49201: Receiving Update with confirmed SDP before 200 ok caused the phone
to drop the outgoing call
49233 Incoming call line key animation is shown even after ending the call at
far end when the phone is initiating conference or transfer.
49237 SoundPoint IP601: One-way audio when changing termination mode
during call waiting when callWaiting.ring="ring" is set.
49256: VVX 1500: If the micro-browser tries to access a URL longer than 54
characters the phone may re-boot or lock-up.
49281: IP7000/HDX integration: When the IP7000 is used to adjust the volume
this may cause the HDX volume level to be reduced to 0.
49287: SUBSCRIBE terminate causes BLF labels to disappear for 2~4 seconds
49323: VVX 1500 reboots after lifting handset while in an empty call list
49402: Race condition when you seize one SCA line and then resume a held
call on another SCA before the line seize completes
49533: Incorrect UDP checksum in DHCP Decline message
49599: BLF: Attendant phone does not update 1/x widget when BLF monitored
local Call Forward behavior.
If set to 1 or Null, the phone will perform
local Call Forward behavior on all calls
received.
sip
added
voIpProt.SIP.tcpFastFailover
If set to 1, failover occurs based on the
values of
reg.x.server.y.retryMaxCount
voIpProt.server.x.retryTimeOut.
If set to 0, use old behavior.
If reg.x.tcpFastFailover is Null,
this attribute is checked.
If voIpProt.SIP.tcpFastFailover
is Null, then this feature is disabled.
If both attributes are set, the value of
reg.x.tcpFastFailover takes
precedence.
sip
changed
voice.gain.tx.digital.headset.IP_430
Changed from 10 to 6
sip
changed
voice.headset.txag.adjust.IP_430
Changed from 39 to 21
sip
changed
dir.corp.pageSize
Changed from 16 to 32
sip
changed
dir.corp.cacheSize
Changed from 64 to 128
sip
added
dir.corp.leg.pageSize
pageSize applied to LDAP queries on
SoundPoint IP 301, 501, 600 and 601
phones. Range is 8 to 64. Default
value is 8
sip
added
dir.corp.leg.cacheSize
cacheSize applied to LDAP queries
on SoundPoint IP 301, 501, 600 and
601 phones. Range is 32 to 256
Default value is 32.
sip
added
dir.corp.sortControl
Controls how client makes queries
and does it sort entries locally. It
should not be used by customers.
If set to 0 or Null, leave sorting as
negotiated between client and
server.
If set to 1, force "non-sorting"
Queries (Not recommended due to
possible performance issues)
sip
added
dir.corp.autoquerySubmitTimeout
To control if there is a timeout after
the user stops entering characters
in the quick search and, if there is,
how long the timeout is.
If set to 0, there is not (disabled).
sip
added
dir.corp.vlv.allow
A flag to determine whether or not
VLV queries can be made if the
LDAP server supports VLV.
If set to 0, VLV queries are disabled.
If set to 1 or Null, VLV queries are
enabled.
The list of attributes (in the exact
order) to be used by the LDAP
server when indexing.
sip
added
dir.corp.attribute.x.searchable
A flag to determine if the attribute is
searchable through quick search.
This flag applies for x = 2 or greater.
If set to 0 or Null, quick search on
this attribute is disabled.
If set to 1, quick search on this
attribute is enabled.
sip
changed
ind.gi.IP_400.6.physW
Changed from 10 to 0
sip
changed
ind.gi.IP_400.6.physH
Changed from 10 to 0
sip
added
pnet.remoteCall.localDialtone
0=no DialTone played when IP 7000
makes an outgoing POTS call on
HDX
1=Play DialTone when IP 7000
makes an outgoing POTS call on
HDX
Default=0
sip
aded
pnet.remoteCall.callProgAtten
Attenuation (in dB) applied to tones
played by the IP 7000 for POTS calls
on HDX when HDX is the active
speaker.
Range -60 to 0; default=-15
2.17 Version 3.1.2 B
2.17.1 Added or Changed Features
Add Support for the VVX 1500 product.
2.17.2 Removed Features
None.
2.17.3 Corrections
None.
2.17.4 Configuration File Parameter Changes
Several parameters added for the VVX 1500 product. See Addendum to SIP 3.1
Administrator‟s Guide for VVX 1500 for details.
2.18 Version 3.1.2
2.18.1 Added or Changed Features
34787: Add Support for ACD Call Center Agent functionality using the „Feature
Synchronization‟ method. See Technical Bulletin 34787 for details.
38442: Add support for multiple NTP servers via DHCP Options 42 or 4 or DNS
SRV or A records.
44612:License file should be provisioned along with configuration files at
application startup.
45233: Implement a „scrolling status bar‟ on phones to match the capability on
the SoundPoint IP 450. This feature applies to all phones except SoundPoint
IP 301.
45460: Add “Quick Set-Up” option. See Technical Bulletin 45460 for details.
45795: Change "Browse Files" to "Browse Recordings" in USB Device menu
46270: Remove DHCP timeout menu option from UI
46631: XML API: Softkeys don't allow for having multiple submit buttons on
the page containing items list
46758: Modify 000000000000.cfg to reference the Configuration File White
Paper
47128: Lifting the handset whilst a BLF monitored line is ringing should seize
a line not answer the remote call. Quick Tip 37381 (see Section 4) has been
updated with to reflect this change.
47309: BLF indicator for a monitored phone should flash when the monitoring
phone calls the monitored phone.
2.18.2 Removed Features
None.
2.18.3 Corrections
25666: 1/A/a not visible when editing some items on SoundPoint IP301.
42425: XML API: Two browser links highlighted after scrolling up a page in a
certain scenario.
43484: CMR/P: Recording does not happen if started while call was on hold
and then resumed.
44271: 200 Response to Cancel is not matched, such that retransmission of
Cancel continues.
44681: SIP 3.0.0 – 3.1.1 Releases: An internal line registration error could
occur if the phone was unable to reach its provisioning server on boot up.
This could result in the phone displaying “Service Unavailable” when the
associated line key was selected.
44727: Microbrowser may display overlapped text if multiple spaces are
included in the page.
45080: Line-seize behavior incorrect for speed-dial when
46767: Configuration parameters bg.gray.selection are repeated in sip.cfg
46807: XML API: Ringer volume adjust tone is repeated every 5s in certain play
URI scenarios
46808: BLF: The 2nd and 3rd Expansion Modules may not work when IP601
monitors 47 BLF lines
46812: XML API: SoundStation IP4000 and IP6000 reboot when attempting to
execute the URI key:line2
46831: Phone locked up with "Reboot initiated" on the display, when it
received corrupted JPEG data.
46843: Using TCP as the transport and BLF line monitoring: An attendant in an
active call cannot perform a directed call pick-up on a remote ringing line.
46858: SoundStation IP 7000 may reboot/freeze if the TRANSFER and CANCEL
soft-keys are pressed in rapid succession.
46861: Call appearance is sometimes missing when a conference is split
during ringback on shared line
46939: Digest Authentication fails on first file in the CONFIG_FILES list with a
certain configuration.
46968: SIP "auth-int" digest authentication mode does not work.
46978: EFK: Configurable soft keys cannot call functions unless at least one
valid efklist entry is present
47083: SoundStation IP 4000: Phone does not send a register request when
parameters qos.ip.rtp.dscp and qos.ip.callControl.dscp are set to a different
value between 0 and 60
47110: SoundStation IP 7000: Enter user password in Advanced menu, phone
goes to Admin menu instead of User menu
47163: 603 Decline sent instead of 486 on DND
47185: In some scenarios, Directed Call-Pickup via BLF drops call and leaves
phone UI in a strange state.
47262: Microbrowser URL in configuration file is not recognized if it is
preceded by spaces
47310: Going on-hook on the handset of the BLF attendant during incoming
call to a BLF monitored line initiates a BLF Call-Pickup.
47345: If call.stickyAutoLineSeize=”1”; In some scenarios, initiating a call
whilst a BLF monitored phone is in the Alerting state may cause the phone to
lock-up.
47450: Port 17185 is open, presenting a security risk
47500: If call.stickyAutoLineSeize=”1”; Active call is not placed on hold when
another call is initiated by a BLF/Speed-dial key.
47530: Using a BLF or Speed Dial key for a Transfer operation does not work.
Add Support for SoundStation IP 7000 integration with HDX Video systems.
This feature requires BootROM 4.1.2
41705: Revise error message, when USB drive is plugged into an IP650/670
and is not supported, to direct phone user to Polycom support web-site.
45411: Change hands-free volume control to give user improved volume level
adjustment capability.
45736: “Reset Device Settings” Menu Option will clear log files on the phone.
45969: Add a menu option to enable/disable headset echo cancellation.
46131: SoundPoint IP 450: Phone does not flash Time and Date when time
server is not configured
2.20.2 Removed Features
None.
2.20.3 Corrections
27694: Interdigit interval of DTMF signal is less than "tone.dtmf.offTime"
setting
30380: In some situations the MWI state is not cleared when all voice msgs on
the phone are deleted.
34586: Phone redials incorrect number after cancelling transfer or conference
41615: Idle display animation will not appear unless phone is used in some
way if the .bmp image only completes downloading after the phone has
booted to the idle screen.
42233: Phone does not attempt Digest Authentication after redirect
43408: BLA line status not updated correctly with a particular signaling timing
scenario.
44099: If attempting to perform a Barge-In on an SCA and the INVITE gets a
403 Forbidden the call no longer shows as active on the phone that tried to
Barge-In
44319: SoundStation IP 6000 and 7000 phones do not use exponential back-off
for TCP retransmissions
44728: Call is not automatically resumed when pressing Cancel soft key after
pressing "URL"
44784: The To-Tag should not be included in an INVITE after a 401 challenge
45039: Unnecessary Refer is sent by phone as it is being blind transferred to a
46888: The phone erroneously sends G.711 mu-law audio with zero SSRC field
regardless of negotiated codec after a conference leg is resumed, a call held
by the far end is resumed, or a remotely held call on a shared/bridged line is
resumed.
2.23 Version 3.1.0 (Limited Distribution; build-id 3.1.0.0073)
This version should be replaced by 3.1.0RevB
2.23.1 Added or Changed Features
22971: Phone should re-register after changing auth parameters.
26010: Add support for Music On Hold (per IETF draft-worley-service-example-
01)
26765: Phone does not handle forked INVITE properly.
29788: Ensure transfer and call termination behavior is robust against
predictable failure modes
30210: Phone should be able to upload a 'tech-support' information dump
31171: Provide New Call soft key when alerting call appearance is in focus
31556: EFK: Add ability to configure Telephony Soft-Keys
32534: Allow on-hook dialing during the alerting state
32757: XML API: Make Micro-browser soft-keys configurable from Server
33428: Exit should exit, Back should take you back
33479: When entering 0 and 00 as speed dial number and saving, phone
should display error message saying invalid Speed Dial number.
33481: Phone should warn if user tries to enter duplicate Speed Dial
34248: Location of Transfer and Conference soft key should not change during
Transfer and Conference process
34364: Add GeoTrust to the built in trusted CA list
37592: Add configuration to give 'dead air' when phone goes off-hook
37644: Limit the number of conference groups to one on SoundStation IP 7000
38022: XML API: Support for asynchronous HTTP URL Push and HTTP POST
to the microbrowser
38032: XML API extensions for application support of telephony functions and
telephony integration
38286: Add support for Plantronics electronic hook switch. This feature
requires BootROM 4.1.0 or newer to operate.
38585: EFK: Add support for enhanced soft key (ESK) capability
38741: EFK: Add the ability to specify a HTTP or HTTPS URL to be loaded by
the microbrowser
38882: Update default list of trusted CAs on the phone
39145: Include Diversion Header Information in the caller-id display
39146: Add ability for the phone to display contents of the SIP warning field to
39647: On registration failure (TCPOnly) phone waits 30-60 seconds for retry
39666: Improve directory configuration parameters – see Administrator‟s
Guide for details.
39821: Add label field to local contact directory
40000: EFK: Add ability to invoke internal key functions via the macro engine
40265: Hide SAS-VP Provisioning Option from the User Interface
40278: SIP stack Tx support of Accept-Language
40341: XML API: Play API - audio file to be downloaded from the HTTP server
and played using the phones speaker.
40431: CMR/P: Add support for USB flash drives larger than 2GB on
SoundPoint IP 650/670 phones.
40543: DTMF dialing will process "," character as 2 sec. pause
40559: When phone is rebooted, it should first deregister before starting
reboot process
40978: EFK: Ensure that all soft key functions can be mapped to hard keys
41016: Add Slovenian to the list of languages supported by certain
SoundPoint/SoundStation IP Phones
41017: Add Polish to the list of languages supported by certain
SoundPoint/SoundStation IP Phones
41050: Enhanced BLF: Add indication of remote phone ringing to Dialog
Package BLF implementation
41161: Add decode support for JPEG image format on SoundStation IP 6000
and 7000 phones.
41177: Add configuration to control whether name or number comes first in
caller-id
41217: Show Diversion Header Information in the caller-id display
41264: Associate key colors with background bitmaps
41366: Update phone UI and Administrator Documents to properly reference
'CDP'
41622: Enhanced BLF: BLF Dialog Handling in SIP Stack
41629: Enhanced BLF: BLF call appearance UI changes
41928: EFK: Remove License requirement from EFK feature
42812: Add EFK support to SoundPoint IP 670
42979: CMR/P: Increase recording buffer size to accommodate flash drives
larger than 2GB
42980: CMR/P: Reject user attempts to perform USB operations while another
operation is still in progress, to support large flash drives.
42982: CMR/P: Add UI icon to show when USB drive is busy, to help user
avoid accidentally removing the drive before an operation finishes
43144: Remove CFS restriction on SSAWC
44546: Set Handset AEC and AES to „on‟ in default configuration files to avoid
handset echo issues.
44740: SoundStation IP 7000: Call lists do not display sip: prefix for URL
dialed calls.
45222: Reduce the default maximum memory size for tones from 600kbytes to
300kbytes to avoid memory issues on SoundPoint IP 320, 330, and 430
products. See Tech Bulletin TB35704 for details on managing the memory
usage on phones.
2.23.2 Removed Features
None.
2.23.3 Corrections
24740: Not all SIP header compact form supported
29946: Log files are not uploaded if an Apache 2.0.X boot server requires
authentication
34586: Phone redials incorrect number after cancelling transfer or conference
in a certain scenario.
35315: URL dialing fails, when shared line is in unregistered state.
35766: Phone locks up after receiving MWI due to extra space in config
36060: nonVolatile.maxSize does not set the contact limit
36728: MWI Caching across re-boots does not work as expected
36770: In ring type menu, ring gets played twice if the wav file is of more than
300kb.
36782: Pressing any digit key should close the pop-up volume control widget.
36933: Menu should not time out when custom certificate fingerprint is being
displayed and user input is expected.
37173: Charge-For-Software: Features not immediately deactivated upon
37924: Peer-to-peer presence: More soft key appears in Buddy Status menu
when there are no more soft keys to display.
38284: Volume adjust -- text labels along with volume bar are incorrect in
some scenarios.
38403: RFC2543 Hold cannot be correctly set using phone's menu and web
Configuration
38452: Press and hold line key, assigning the in-focus entry to that speed dial
key does not work correctly
38548: Typing some value in the "Send message to:" field and exiting causes
problem when "Instant Messages" is re-selected.
38610: Burst of ring tone happens before ring back when call is placed for the
2nd time after the 1st call is dropped.
38631: Go to Directory menu, down scrolling icon does not display until down
arrow key is pressed if contact does not have last/first name
38633: [Corporate Directory] When there are no records in Corporate Directory
menu, Search soft key should not display
38636: CMR/P: Wav file cannot be opened when consultation call (of
Conference) is on hold.
38798: Operation of menus using the 'Back' softkey are confusing
39022: Transfer and Conference softkeys are still available on
IP650/IP550/IP301/IP4000 after maximum number of outgoing calls are made
from these phones.
39208: Content Type Header field not handled properly in Microbrowser
39317: Call cannot be resumed when reINVITE is given a 404 error
39533: Malicious connection to TCP port 5060 may cause phone to reboot
39546: [Presence]: phone should not send Presence SUBSCRIBE signaling
when pres.reg = invalid line number
39553: Corporate Directory: when DNS record timeouts, Corp Dir does not
honour TTL and sends a new DNS query
39598: VQMon: use of partition byte count (magic number) to detect SID/CNG
is too small - use buffer flags instead
39623: Headset: Headset icon (active path icon) disappears during call in a
certain scenario on the SoundPoint IP 430 phone.
39642: SoundStation IP 6000 and 7000 products reply to IP packets of
unknown protocol with ICMP messages
39788: SoundPoint IP 501, 601: Phone should not play incoming rtp when
offered recvonly stream.
39935: Users of the IP650 hands free complain that sometimes, the phone
goes dead silent and they wonder if the far-end is still on the line
40586: SoundStation IP 7000 : Phone's UI does not display ''date and time'' in
the call appearance screen during multiple calls
40660: + being „escaped‟ as %2B in INVITE URI
40664: To establish a 2nd call using speaker key while the first call is on hold,
one has to press the speaker key twice.
40716: CMR/P: Renaming the new wav file to an already existing old wav file
should be prohibited. Currently, this failure replaces the new file completely
(content, length, size) with old file.
40718: CMR/P: Rename screen: (1) Title is incomplete. (2) Encoding soft key
appears after second press of 1/A/a soft key.
40804: CMR/P: When new call arrives while user is in the audio player screen
but not playing audio, incorrect softkeys are displayed
40831: Corporate Directory: Page and Cache size parameters should be
configurable.
40862: Wrong soft key displayed while transferring a url call and selecting
blind
40898: Usage bar shows behind customer bitmap display
40945: Pressing DND feature during hot dial creates problem with new call
establishment.
41002: When entering contact directory entry, there is no soft key (1/A/a) to
change number/lower case/upper case
41034: CMR/P: No audio in Jabra 9350 headset when wav file is played through
headset mode, though the visual indicators show it in "Playing" state.
41173: Japanese XML dictionary needs a review
41184: SoundStation IP 7000: Wrong Date Time format when you select
Japanese language
41186: SoundStation IP 7000: Date Time format is wrong on the
Placed/Received Calls info when Japanese Language is selected.
41364: Phones does not honor MIME type for telephone event in SDP Answer
41448: Phone stops sending DTMF in a certain scenario
41700: RTP does not go to correct destination following reINVITE
42252: Configuring VLAN discovery does not incur a restart
42261: Phone will not search sub containers in the corporate directory
42749: Phone connects to LDAP server, but does not return records
42792: Media Attribute missing in Hold ReINVITE when SRTP is enabled.
42841: Echo is experienced when calling IP 650 to IP 650 using G.722 HD at full
43014: call.stickyAutoLineSeize is not working correctly when a second call is
initiated from a speed dial.
43121: safeReconfig on SoundStation IP 4000 results in the phone rebooting.
43360: Phone sends a „terminated‟ notify with two different dialogs for the
same call
43513: SoundPoint IP 650 experiencing Echo at full volume on handset
43745: French XML Dictionary needs updating
44066: Ringer diminishes on some phones over time and stops working
44164: SoundPoint IP 320 does not respond to UPDATE when sent more than
9 seconds after INVITE
44223: SoundStation IP 7000: # key behaves as if pressing the “1/A/a “ soft key
44324: Feature key remapping does not always work
44029: When ANALOG HEADSET MODE is set to JABRA mode, there is no
audio call waiting tone.
44066: Ringer (including call waiting tone) volume diminishes on some phones
over time and stops being audible.
44413: Speed dial labels on line keys are switched from first, last to last first.
44423: Speed dial entries on 650s are coming up “URL Call Disabled”
44509: SoundPoint IP 600/601: Transferring and originating calls generates
“URL Call Disabled” message.
44520: Phone is generating aan unexpected NOTIFY on an incoming call which
puts the BLA status out of sync.
44763: Phones ignoring DNS SRV records response from Session Border
Controller in certain scenario
45093: SoundStation IP4000 and 6000 have no way to delete or backspace on
the Password entry screen.
45118: Digest authentication for SIP transactions fail when “digest” token is in
lower-case characters
45198: Dialing EFK macros from speed dial key does not work if URL dialing is
If set to 1, forces the phone to wait for
200 OK response when receiving a
TRYING notify.
If set to 0 or Null, this is old behavior.
sip
added
voIpProt.SIP.strictUserValidation
If set to 1, forces the phone to match
user portion of signaling exactly.
If set to 0 or Null, phone will use first
registration if the user part does not
match any registration.
sip
added
voIpProt.SIP.lineSeize.retries
Controls the number of times the
phone will retry a notify when
attempting to seize a line (BLA).
sip
added
voIpProt.SIP.header.diversion.enable
If set to 1, the diversion header is
displayed if received.
If set to 0 or Null, the diversion
header is not displayed.
sip
added
voIpProt.SIP.header.list.useFirst
If set to 1 or Null, the first diversion
header is displayed.
If set to 0, the last diversion header is
displayed.
sip
added
voIpProt.SIP.header.warning.codes.accept
A list of accepted warning codes.
If set to Null, all codes are accepted.
Only codes between 300 and 399 are
supported.
sip
added
voIpProt.SIP.header.warning.enable
If set to 1, the warning header is
displayed if received.
If set to 0 or Null, the warning header
is not displayed.
sip
added
voIpProt.SIP.musicOnHold.uri
A URI that provides the media stream
to play for the remote party on hold.
If reg.x.musicOnHold is set to Null,
this attribute is checked.
sip
added
lcl.ml.lang.tags.x
The format is:
• The first two letters are the ISO-639
language abbreviation.
• The next two letters are the ISO3166 country code.
• The next two letters are the ISO-639
language abbreviation.
• The remainder of the string is the
preference level for the display of the
language, or English if the language
is not available
sip
added
up.numberFirst CID
If set to 0 or Null, caller ID display will
show caller‟s name first.
If set to 1, caller ID display will show
caller‟s number first.
sip
changed
saf.1
The default value is Null. To allow the
SoundPoint IP welcome sound to be
played on reboots and restarts, set to
SoundPointIPWelcome.wav
The maximum number of entries
requested from the corporate
directory server with each query.
sip
added
dir.corp.cacheSize
The maximum number of entries
that can be cached locally on the
phone.
sip
added
dir.corp.scope
Type of search.
If set to “one”, a search of the level
one below the baseDN is performed.
If set to “sub” or Null, a recursive
search (of all levels below the
baseDN) is performed.
If set to “base”, a search at the
baseDN level is performed.
sip
changed
voice.ns.hs.enable
The default value is enabled (1).
sip
changed
res.quotas.1.value
The default value is 300KB for tones.
sip
added
apps.telNotification.URL
The URL to which the phone sends
notifications of specified events. The
protocol used can be either HTTP or
HTTPS.
sip
added
apps.telNotification.incomingEvent
If set to 0, incoming call notification is
disabled.
If set to 1, incoming call notification is
enabled.
sip
added
apps.telNotification.outgoingEvent
If set to 0, outgoing call notification is
disabled.
If set to 1, outgoing call notification is
enabled.
sip
added
apps.telNotification.offhookEvent
If set to 0, offhook notification is
disabled.
If set to 1, offhook notification is
enabled
sip
added
apps.telNotification.onhookEvent
If set to 0, onhook notification is
disabled.
If set to 1, onhook notification is
enabled
sip
added
apps.statePolling.URL
The URL to which the phone sends
call processing state/device/network
information. The protocol used can be
either HTTP or HTTPS
sip
added
apps.statePolling.username
The user name to access the state
polling URL.
sip
added
apps.statePolling.password
The password to access the state
polling URL.
sip
added
apps.push.messageType
Select the allowable push priority
messages on
phone.
sip
added
apps.push.serverRootURL
The relative URL (received from
HTTP URL Push message) is
appended to the application server
root URL and the resultant URL is
sent to the Microbrowser.
This is the text displayed with the soft
key.
If set to Null, the label to display is
determined as follows:
• If the soft key is mapped to a
enhanced feature key macro, the
label of the enhanced feature key
macro will be used.
• If the soft key is mapped to a speed
dial, the label of the corresponding
directory entry will be used. If this
label does not exist as well and the
directory entry is an enhanced feature
key macro, then the label of the
enhanced feature key macro will be
used.
• If the soft key is mapped to chained
actions, only the first one is
considered for label, using the rules
above.
• If no labels are found after the
above steps, the soft key label will be
blank.
sip
added
softkey.x.action
The same syntax as the enhanced
feature key action.
sip
added
softkey.x.enable
If set to 0 or Null, the soft key is
disabled.
If set to 1, the soft key is enabled.
sip
added
softkey.x.precede
If set to 0 or Null, the soft key
replaces any empty space from the
leftmost position.
If set to 1, the soft key is displayed
before the first standard soft key.
sip
added
softkey.x.use.idle
If set to 0 or Null, the soft key is not
displayed in the idle state.
If set to 1, the soft key is displayed in
the idle state.
sip
added
softkey.x.use.active
If set to 0 or Null, the soft key is not
displayed in the active call state.
If set to 1, the soft key is displayed in
the active call state.
sip
added
softkey.x.use.alerting
If set to 0 or Null, the soft key is not
displayed in the alerting state.
If set to 1, the soft key is displayed in
the alerting state.
sip
added
softkey.x.use.dialtone
If set to 0 or Null, the soft key is not
displayed in the dialtone state.
If set to 1, the soft key is displayed in
the dialtone state.
sip
added
softkey.x.use.proceeding
If set to 0 or Null, the soft key is not
displayed in the proceeding state.
If set to 1, the soft key is displayed in
the proceeding state.
sip
added
softkey.x.use.setup
If set to 0 or Null, the soft key is not
displayed in the setup state.
If set to 1, the soft key is displayed in
the setup state.
If set to 0 or Null, the soft key is not
displayed in the hold state.
If set to 1, the soft key is displayed in
the hold state.
sip
added
softkey.feature.newcall
If set to 0, the New Call soft key is not
displayed when there is another way
to place a call.
If set to 1 or Null, the New Call soft
key is displayed.
sip
added
softkey.feature.endcall
If set to 0, the End Call soft key is not
displayed.
If set to 1 or Null, the EndCall soft key
is displayed.
sip
added
softkey.feature.split
If set to 0, the Split soft key is not
displayed.
If set to 1 or Null, the Split soft key is
displayed.
sip
added
softkey.feature.join
If set to 0, the Join soft key is not
displayed.
If set to 1 or Null, the Join soft key is
displayed.
sip
added
softkey.feature.forward
If set to 0, the Forward soft key is not
displayed.
If set to 1 or Null, the Forward soft
key is displayed.
sip
added
softkey.feature.directories
If set to Null, the Dir soft key is
displayed on the SoundPoint IP
320/330 phone, but not on any other
phone.
If set to 0, the Dir soft key is not
displayed on any phone.
If set to 1, the Dir soft key is displayed
on all phones as follows:
• In the idle state, it is displayed after
the New Call and Callers soft keys.
• In the dialtone state, it is displayed
after the End Call and Callers soft
keys.
• During a conference or transfer, it is
displayed after the Callers and
Cancel soft keys.
sip
added
softkey.feature.callers
If set to Null, the Callers soft key is
displayed on the SoundPoint IP
320/330 phone, but not on any other
phone.
If set to 0, the Callers soft key is not
displayed on any phone.
If set to 1, the Callers soft key is
displayed on all phones as follows:
• In the idle state, it is displayed after
the New Call soft key and before the
Dir soft key.
• In the dialtone state, it is displayed
after the End Call soft key and before
the Dir soft key.
• During a conference or transfer, it is
displayed before the Cancel soft key.
If set to 0, the MyStatus soft key is
not displayed.
If set to 1 or Null, the MyStatus soft
key is displayed.
sip
added
softkey.feature.buddies
If set to 0, the Buddies soft key is not
displayed.
If set to 1 or Null, the Buddies soft key
is displayed.
sip
added
softkey.feature.basicCallManagement.redu
ndant
If set to 0 and the phone has hard
keys mapped for Hold, Transfer, and
Conference functions (all must be
mapped), all of these soft keys are
not displayed.
If set to 1 or Null, all of these soft
keys are displayed.
phone1
added
reg.x.strictLineSeize
If set to 1, forces phone to wait for
200 OK on registration x when
receiving a TRYING notify.
If set to 0 or Null, this is old behavior.
If this parameter is Null,
voIpProt.SIP.strictLineSeize is
checked.
If both parameters are set, this
parameter takes precedence.
phone1
added
reg.x.musicOnHold.uri
A URI that provides the media stream
to play for the remote party on hold.
When present, and if
reg.x.musicOnHold is not Null, this
attribute overrides the global Music
on Hold defined in the sip.cfg
configuration file.
phone1
added
attendant.ringType
The ring tone to play when a BLF
dialog is in the offering state.
Permitted values are 1 to 22. The
default is Null.
2.24 Version 3.0.4
Note that Version 3.0.4 was released after SIP 3.1.0, so it should not be assumed that the
changes in SIP 3.0.4 also apply to SIP 3.1.0.
2.24.1 Added or Changed Features
44546: Set Handset AEC and AES to „on‟ in default configuration files to avoid
handset echo issues.
45411: Adjust Speaker phone (Hands Free) volume control for better user
experience.
2.24.2 Removed Features
None.
2.24.3 Corrections
43264: Phone is not able to answer calls due to duplicate INVITEs with same