Release Notes
SIP Application
®
SoundPoint® IP, SoundStation® IP, and VVX
Version 3.2.4B
11 February 2011
Part Number 3804-11530-324B
Copyright © 2011 Polycom, Inc. All rights reserved.
Release Notes - SIP Application
Trademark Information
© 2011, Polycom, Inc. All rights reserved. POLYCOM®, the Polycom "Triangles" logo and
the names and marks associated with Polycom's products are trademarks and/or service
marks of Polycom, Inc. and are registered and/or common law marks in the United States
and various other countries. All other trademarks are property of their respective owners.
No portion hereof may be reproduced or transmitted in any form or by any means, for any
purpose other than the recipient's personal use, without the express written permission of
Polycom.
Copyright © 2011 Polycom, Inc. All rights reserved.
Release Notes - SIP Application Table of Contents
Table of Contents
1. GENERAL ................................................................................................................................... 1
1.1 IMPORTANT NOTES ................................................................................................................ 1
1.2 FEATURE LICENSE AND PLATFORM LIMITATIONS .................................................................. 2
1.3 SYSTEM REQUIREMENTS ........................................................................................................ 4
1.4 DISTRIBUTION FILES .............................................................................................................. 5
1.4.1 Release using individual (split) files ................................................................................. 5
1.4.2 Release using Combined Image ........................................................................................ 6
2. CHANGES ................................................................................................................................... 8
2.1 VERSION 3.2.4B ..................................................................................................................... 8
2.1.1 Added or Changed Features ............................................................................................. 8
2.1.2 Removed Features ............................................................................................................. 8
2.1.3 Corrections ....................................................................................................................... 8
2.2 VERSION 3.2.4 ....................................................................................................................... 8
2.2.1 Added or Changed Features ............................................................................................. 8
2.2.2 Removed Features ............................................................................................................. 8
2.2.3 Corrections ....................................................................................................................... 8
2.3 VERSION 3.2.3 ....................................................................................................................... 8
2.3.1 Added or Changed Features ............................................................................................. 8
2.3.2 Removed Features ............................................................................................................. 9
2.3.3 Corrections ....................................................................................................................... 9
2.3.4 Configuration File Parameter Changes ......................................................................... 12
2.4 VERSION 3.2.2 ..................................................................................................................... 14
2.4.1 Added or Changed Features ........................................................................................... 14
2.4.2 Removed Features ........................................................................................................... 14
2.4.3 Corrections ..................................................................................................................... 14
2.4.4 Configuration File Parameter Changes ......................................................................... 18
2.5 VERSION 3.2.1 B .................................................................................................................. 20
2.5.1 Added or Changed Features ........................................................................................... 20
2.5.2 Removed Features ........................................................................................................... 20
2.5.3 Corrections ..................................................................................................................... 20
2.5.4 Configuration File Parameter Changes ......................................................................... 20
2.6 VERSION 3.2.1 ..................................................................................................................... 21
2.6.1 Added or Changed Features ........................................................................................... 21
2.6.2 Removed Features ........................................................................................................... 21
2.6.3 Corrections ..................................................................................................................... 21
2.6.4 Configuration File Parameter Changes ......................................................................... 21
2.7 VERSION 3.2.0 ..................................................................................................................... 22
2.7.1 Added or Changed Features ........................................................................................... 22
2.7.2 Removed Features ........................................................................................................... 24
2.7.3 Corrections ..................................................................................................................... 24
2.7.4 Configuration File Parameter Changes ......................................................................... 34
2.8 VERSION 3.1.6 ..................................................................................................................... 41
2.8.1 Added or Changed Features ........................................................................................... 41
2.8.2 Removed Features ........................................................................................................... 41
Copyright © 2011 Polycom, Inc. Page i
Release Notes - SIP Application Table of Contents
2.8.3 Corrections ..................................................................................................................... 41
2.8.4 Configuration File Parameter Changes ......................................................................... 41
2.9 VERSION 3.1.5 (LIMITED DISTRIBUTION) ............................................................................. 41
2.9.1 Added or Changed Features ........................................................................................... 41
2.9.2 Removed Features ........................................................................................................... 41
2.9.3 Corrections ..................................................................................................................... 41
2.9.4 Configuration File Parameter Changes ......................................................................... 41
2.10 VERSION 3.1.4 ..................................................................................................................... 42
2.10.1 Added or Changed Features ....................................................................................... 42
2.10.2 Removed Features ....................................................................................................... 42
2.10.3 Corrections ................................................................................................................. 42
2.10.1 Configuration File Parameter Changes ..................................................................... 42
2.11 VERSION 3.1.3 C .................................................................................................................. 42
2.11.1 Added or Changed Features ....................................................................................... 42
2.11.2 Removed Features ....................................................................................................... 42
2.11.3 Corrections ................................................................................................................. 42
2.11.4 Configuration File Parameter Changes ..................................................................... 43
2.12 VERSION 3.1.3 B .................................................................................................................. 43
2.12.1 Added or Changed Features ....................................................................................... 43
2.12.2 Removed Features ....................................................................................................... 43
2.12.3 Corrections ................................................................................................................. 43
2.12.4 Configuration File Parameter Changes ..................................................................... 43
2.13 VERSION 3.1.3 (LIMITED RELEASE – VERSION 3.1.3.0336 ) ................................................. 43
2.13.1 Added or Changed Features ....................................................................................... 43
2.13.2 Removed Features ....................................................................................................... 44
2.13.3 Corrections ................................................................................................................. 44
2.13.4 Configuration File Parameter Changes ..................................................................... 48
2.14 VERSION 3.1.2 B .................................................................................................................. 49
2.14.1 Added or Changed Features ....................................................................................... 49
2.14.2 Removed Features ....................................................................................................... 49
2.14.3 Corrections ................................................................................................................. 49
2.14.4 Configuration File Parameter Changes ..................................................................... 49
2.15 VERSION 3.1.2 ..................................................................................................................... 49
2.15.1 Added or Changed Features ....................................................................................... 49
2.15.2 Removed Features ....................................................................................................... 50
2.15.3 Corrections ................................................................................................................. 50
2.15.4 Configuration File Parameter Changes ..................................................................... 54
2.16 VERSION 3.1.1 B .................................................................................................................. 54
2.16.1 Added or Changed Features ....................................................................................... 54
2.16.2 Removed Features ....................................................................................................... 54
2.16.3 Corrections ................................................................................................................. 54
2.16.4 Configuration File Parameter Changes ..................................................................... 55
2.17 VERSION 3.1.1 ..................................................................................................................... 55
2.17.1 Added or Changed Features ....................................................................................... 55
2.17.2 Removed Features ....................................................................................................... 55
2.17.3 Corrections ................................................................................................................. 55
2.17.4 Configuration File Parameter Changes ..................................................................... 57
2.18 VERSION 3.1.0 C .................................................................................................................. 57
2.18.1 Added or Changed Features ....................................................................................... 57
Page ii Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Table of Contents
2.18.2 Removed Features ....................................................................................................... 57
2.18.3 Corrections ................................................................................................................. 57
2.18.4 Configuration File Parameter Changes ..................................................................... 58
2.19 VERSION 3.1.0 B .................................................................................................................. 58
2.19.1 Added or Changed Features ....................................................................................... 58
2.19.2 Removed Features ....................................................................................................... 58
2.19.3 Corrections ................................................................................................................. 58
2.19.4 Configuration File Parameter Changes ..................................................................... 58
2.20 VERSION 3.1.0 (LIMITED DISTRIBUTION; BUILD-ID 3.1.0.0073) ........................................... 59
2.20.1 Added or Changed Features ....................................................................................... 59
2.20.2 Removed Features ....................................................................................................... 61
2.20.3 Corrections ................................................................................................................. 61
2.20.4 Configuration File Parameter Changes ..................................................................... 66
2.21 VERSION 3.0.4 ..................................................................................................................... 70
2.21.1 Added or Changed Features ....................................................................................... 70
2.21.2 Removed Features ....................................................................................................... 70
2.21.3 Corrections ................................................................................................................. 70
2.21.4 Configuration File Parameter Changes ..................................................................... 71
2.22 VERSION 3.0.3 B .................................................................................................................. 71
2.22.1 Added or Changed Features ....................................................................................... 71
2.22.2 Removed Features ....................................................................................................... 71
2.22.3 Corrections ................................................................................................................. 72
2.22.4 Configuration File Parameter Changes ..................................................................... 72
2.23 VERSION 3.0.3 ..................................................................................................................... 72
2.23.1 Added or Changed Features ....................................................................................... 72
2.23.2 Removed Features ....................................................................................................... 72
2.23.3 Corrections ................................................................................................................. 72
2.23.4 Configuration File Parameter Changes ..................................................................... 74
2.24 VERSION 3.0.2 C .................................................................................................................. 74
2.24.1 Added or Changed Features ....................................................................................... 74
2.24.2 Removed Features ....................................................................................................... 74
2.24.3 Corrections ................................................................................................................. 74
2.24.4 Configuration File Parameter Changes ..................................................................... 74
2.25 VERSION 3.0.2 B (LIMITED RELEASE – BUILD-ID 3.0.2.0917) .............................................. 74
2.25.1 Added or Changed Features ....................................................................................... 74
2.25.2 Removed Features ....................................................................................................... 75
2.25.3 Corrections ................................................................................................................. 75
2.25.4 Configuration File Parameter Changes ..................................................................... 77
2.26 VERSION 3.0.1REVB ............................................................................................................ 78
2.26.1 Added or Changed Features ....................................................................................... 78
2.26.2 Removed Features ....................................................................................................... 78
2.26.3 Corrections ................................................................................................................. 78
2.27 VERSION 3.0.1 (LIMITED DISTRIBUTION – BUILD-ID 3.0.1.0032) ......................................... 79
2.27.1 Added or Changed Features ....................................................................................... 79
2.27.2 Removed Features ....................................................................................................... 79
2.27.3 Corrections ................................................................................................................. 79
2.27.4 Configuration File Parameter Changes ..................................................................... 79
2.28 VERSION 3.0.0 ..................................................................................................................... 79
2.28.1 Added or Changed Features ....................................................................................... 79
Copyright © 2011 Polycom, Inc. Page iii
Release Notes - SIP Application Table of Contents
2.28.2 Removed Features ....................................................................................................... 81
2.28.3 Corrections ................................................................................................................. 81
2.28.4 Configuration File Parameter Changes ..................................................................... 84
2.29 VERSION 2.2.2 ..................................................................................................................... 84
2.29.1 Added or Changed Features ....................................................................................... 84
2.29.2 Removed Features ....................................................................................................... 85
2.29.3 Corrections ................................................................................................................. 85
2.29.4 Configuration File Parameter Changes ..................................................................... 86
2.30 VERSION 2.2.1 (LIMITED RELEASE) ..................................................................................... 86
2.30.1 Added or Changed Features ....................................................................................... 86
2.30.2 Removed Features ....................................................................................................... 86
2.30.3 Corrections ................................................................................................................. 86
2.30.4 Configuration File Parameter Changes ..................................................................... 87
2.31 VERSION 2.2.0 ..................................................................................................................... 87
2.31.1 Added or Changed Features ....................................................................................... 87
2.31.2 Removed Features ....................................................................................................... 89
2.31.3 Corrections ................................................................................................................. 89
2.31.4 Configuration File Parameter Changes ..................................................................... 92
2.32 VERSION 2.1.2 ..................................................................................................................... 96
2.32.1 Added or Changed Features ....................................................................................... 96
2.32.2 Removed Features ....................................................................................................... 96
2.32.3 Corrections ................................................................................................................. 96
2.32.4 Configuration File Parameter Changes ..................................................................... 97
2.33 VERSION 2.1.1 C .................................................................................................................. 98
2.33.1 Added or Changed Features ....................................................................................... 98
2.33.2 Removed Features ....................................................................................................... 98
2.33.3 Corrections ................................................................................................................. 98
2.33.4 Configuration File Parameter Changes ..................................................................... 99
2.34 VERSION 2.1.1 ..................................................................................................................... 99
2.34.1 Added or Changed Features ....................................................................................... 99
2.34.2 Removed Features ....................................................................................................... 99
2.34.3 Corrections ................................................................................................................. 99
2.34.4 Configuration File Parameter Changes ................................................................... 101
2.35 VERSION 2.1.0 ................................................................................................................... 102
2.35.1 Added or Changed Features ..................................................................................... 102
2.35.2 Removed Features ..................................................................................................... 103
2.35.3 Corrections ............................................................................................................... 103
2.35.4 Configuration File Parameter Changes ................................................................... 105
3. OUTSTANDING ISSUES ...................................................................................................... 107
4. REFERENCE DOCUMENTS ............................................................................................... 113
Page iv Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application General
1. General
These release notes apply to version 3.2.4B of the SoundPoint IP, SoundStation IP and
VVX SIP application. For more information, refer to the documents listed in Section 4.
1.1 Important Notes
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:
http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/
VoIP_Technical_Bulletins_pub.html.
VVX 1500 products running release SIP 3.2.2 or later CANNOT BE
DOWNGRADED TO EARLIER SIP SOFTWARE OR BOOTROM SOFTWARE.
Upgrading VVX1500 products to release SIP 3.2.2 or later requires a more complex
procedure than is typical. This procedure is documented in technical bulletin
TB53522 Please consult this document before starting the upgrade.
This release does not include support for the SoundPoint IP 300, 301, 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) and SIP 3.1.x release (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://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/
VoIP_Technical_Bulletins_pub.html. The template 000000000000.cfg file included
with this release is set up to facilitate this type of deployment.
SoundStation IP 7000/HDX Integration:
Release SIP 3.2.4 is recommended for SoundStation IP 7000 integration with
Polycom HDX 4000/6000/7000/8000/9000 video systems release HDX 2.6.0,
2.6.0.2, 2.6.1 and 2.6.1.3
The sip.cfg template file included with this release contains language selections in
the „native‟ font for that language. These include fonts that are not supported in certain
XML editors. If the sip.cfg is edited using such an editor the language selections
shown in the Languages menu on the phone may not display correctly. To confirm
whether your editor properly supports these characters, view the language
parameter for languages such as Chinese, Japanese, Korean, Russian – e.g.
lcl.ml.lang.menu.1.label
Documentation on how to enable and use the new features in the SIP 3.2.x release
is included in the Administrator‟s Guide for SIP 3.2.2 (See Section 4 for details on
how to access the document). There is a specific section in this document that
references the major new features in the SIP 3.2 Release.
Copyright © 2011 Polycom, Inc. Page 1
Release Notes - SIP Application General
Feature
IP 320/330
IP 321/331/335
IP 430
IP 450/550/560
IP 650/670
VVX 1500/-C/-D
VQMon
Productivity
License
Productivity
License
Productivity
License
Productivity
License
Productivity
License
Yes (Audio only)
LDAP Directory
Productivity
License
Productivity
License
Productivity
License
Productivity
License
Productivity
License
Yes
Call Recording
No
No
No
No
Productivity
License
Yes (Audio only)
Conference
Management
No
No
No
Productivity
License
Productivity
License
Yes
4-way local
conference
No
No
No
Productivity
License
Productivity
License
No
Electronic
Hookswitch
Yes
Yes
Yes
Yes
Yes
Yes
Enhanced
Feature Keys
Yes
Yes
Yes
Yes
Yes
Yes
Customizable UI
Background
No
No
No
Yes
Yes
Yes
Local SRTP
Conference
Yes
Yes
Yes
Yes
Yes
Yes, with
limitations at high
video bandwidths
Asian Language
No
No
No
Yes
Yes
Yes
Configurable
Soft-Keys
Yes
Yes
Yes
Yes
Yes
Yes
XML API
Yes
Yes
Yes
Yes
Yes
Yes
Enhanced BLF
No
No
No
Yes
Yes
No
Warning Field
Display
Yes Yes
Yes
Yes
Yes
Yes
H.323 Video
No
No
No
No
No
License (pre-
installed on
1500D)
1.2 Feature License and Platform limitations
The following table summarizes several features that require a particular hardware platform
and/or a license key for activation.
SoundPoint IP and Polycom VVX Family of Products (Desktop Phones)
Productivity License – licensed as part of the Productivity Suite
Page 2 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application General
Feature
IP5000
IP 6000
IP 7000
VQMon
Yes
No
No
LDAP
Directory
Productivity
License
Productivity
License
Yes
Call
Recording
No
No
No
Conference
Management
No
No
Yes
4-way local
conference
No
No
No
Electronic
Hookswitch
No
No
No
Enhanced
Feature Keys
No
No
No
Customizable
UI
Background
No
No
No
Local SRTP
Conference
Yes
Yes
Yes
Asian
Language
Yes
Yes
Yes
Configurable
Soft-Keys
No
No
No
XML API
Yes
Yes
Yes
Enhanced
BLF
No
No
No
Warning
Field Display
Yes
Yes
Yes
H.323 Video
No
No
No
SoundStation IP Product Family (Conference Phones)
Copyright © 2011 Polycom, Inc. Page 3
Release Notes - SIP Application General
Platform
BootROM version
SoundPoint IP 320/330
3.2.3RevB or later
SoundPoint IP 321/331
4.1.3 or later
SoundPoint IP 335
4.2.0RevB or later
SoundPoint IP 430
3.1.3 or later
SoundPoint IP 450
4.1.2 or later
SoundPoint IP 550
4.1.0 or later
SoundPoint IP 560
4.1.0 or later
SoundPoint IP 650
4.1.0 or later
SoundPoint IP 670
4.1.1 or later
SoundStation IP 5000
4.2.2 or later
SoundStation IP 6000
4.1.1 or later
SoundStation IP 7000
4.1.1 or later
SoundStation IP 7000
(used with Polycom HDX 4000, 7000,
8000, 9000 video systems)
4.2.2 or later
SoundStation IP 7000 used with HDX
6000 video systems. SIP 3.2.1 Cannot
integrate with HDX until HDX version
supporting integration is announced.
4.2.2 or later
VVX 1500
4.2.2 (NOTE: As of 3.2.2, the SIP and
BootROM are distributed as single
package for VVX1500)
1.3 System Requirements
Although it is not a requirement, it is recommended that BootROM 4.2.3 be used in
conjunction with SIP 3.2.4B.
For details on historical software version support by platform please refer to the “SIP
Downloads Matrix” table accessible from the Polycom Support site at
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
Page 4 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application General
Files
Description
2345-12200-002.sip.ld
2345-12200-005.sip.ld
SIP application executables for SoundPoint IP 320
2345-12360-001.sip.ld
SIP application executables for SoundPoint IP 321
2345-12200-001.sip.ld
2345-12200-004.sip.ld
SIP application executables for SoundPoint IP 330
2345-12365-001.sip.ld
SIP application executables for SoundPoint IP 331
2345-12375-001.sip.ld
SIP application executables for SoundPoint IP 335
2345-11402-001.sip.ld
SIP application executable for SoundPoint IP 430
2345-12450-001.sip.ld
SIP application executable for SoundPoint IP 450
2345-12500-001.sip.ld
SIP application executable for SoundPoint IP 550
2345-12560-001.sip.ld
SIP application executable for SoundPoint IP 560
2345-12600-001.sip.ld
SIP application executable for SoundPoint IP 650
2345-12670-001.sip.ld
SIP application executable for SoundPoint IP 670
3111-30900-001.sip.ld
SIP application executable for SoundStation IP 5000
3111-15600-001.sip.ld
SIP application executable for SoundStation IP 6000
3111-40000-001.sip.ld
SIP application executable for SoundStation IP 7000
2345-17960-001.sip.ld
SIP application executable for VVX 1500
sip.cfg
main core and SIP configuration file
phone1.cfg
example per-phone SIP configuration
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)
1.4 Distribution Files
The SIP 3.2.4B distribution of the SoundPoint / SoundStation IP/VVX SIP application 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 (except the VVX 1500) of the “.sip.ld” files listed below (both
split and combined) is now at revision: 3.2.4.0267
The current build ID for the VVX 1500 “2345-17960-001.sip.ld” file listed below remains
at: 3.2.4.0244
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.
Copyright © 2011 Polycom, Inc. Page 5
Release Notes - SIP Application General
Files
Description
SoundPointIP-dictionary.xml
dictionary files for multilingual support include:
Chinese, China (for IP 450, 550, 560, 650 and IP 5000, 6000, 7000)
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)
Korean, Korea (for IP 450, 550, 560, 650, 670 and IP 5000, 6000,
7000)
Norwegian, Norway
Polish, Poland
Portuguese, Portugal
Russian, Russia
Slovenian, Slovenia
Spanish, Spain
Swedish, Sweden
SoundPointIPWelcome.wav
start up welcome sound effect
Files
Description
sip.ld
Concatenated SIP application executable
sip.cfg
main core and SIP configuration file
phone1.cfg
example per-phone SIP configuration
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)
SoundPointIP-dictionary.xml
dictionary files for multilingual support include:
Chinese, China (for IP 450, 550, 560, 650 and IP 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 6000, 7000)
Korean, Korea (for IP 450, 550, 560, 650, 670 and IP 6000, 7000)
Norwegian, Norway
Polish, Poland
Portuguese, Portugal
Russian, Russia
Slovenian, Slovenia
Spanish, Spain
Swedish, Sweden
1.4.2 Release using Combined Image
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 older than SIP 4.0.0 (e.g. BootROM 3.2.3RevB).
Page 6 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application General
Files
Description
SoundPointIPWelcome.wav
start up welcome sound effect
LoudRing.wav
Loud ringer sound effect
Copyright © 2011 Polycom, Inc. Page 7
Release Notes - SIP Application Changes
2. Changes
2.1 Version 3.2.4B
2.1.1 Added or Changed Features
N/A
2.1.2 Removed Features
N/A
2.1.3 Corrections
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 3.2.4
2.2.1 Added or Changed Features
N/A
2.2.2 Removed Features
N/A
2.2.3 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
unprotected section of flash memory.
2.3 Version 3.2.3
2.3.1 Added or Changed Features
43099: Added support for SoundStation IP5000 Conference Phone.
Page 8 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
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”.
2.3.2 Removed Features
N/A
2.3.3 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
such as when the remote phone is placed on hold.
52787: voIpProt.SIP.requestValidation.x.method="source" does not work with
DNS SRV Static Cache
Copyright © 2011 Polycom, Inc. Page 9
Release Notes - SIP Application Changes
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.
Page 10 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
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.
Copyright © 2011 Polycom, Inc. Page 11
Release Notes - SIP Application Changes
File
Change
Attribute
Old
value
New value
phone1
added
reg.n.server.1.failOver.reRegisterOn
phone1
added
reg.n.server.1.failOver.failBack.mode
phone1
added
reg.n.server.1.failOver.failBack.timeout
phone1
added
reg.n.server.2.failOver.reRegisterOn
phone1
added
reg.n.server.2.failOver.failRegistrationOn
phone1
added
reg.n.server.2.failOver.failBack.mode
phone1
added
reg.n.server.2.failOver.failBack.timeout
phone1
added
reg.n.outboundProxy.failOver.reRegisterOn
phone1
added
reg.n.outboundProxy.failOver.failRegistrationOn
phone1
added
reg.n.outboundProxy.failOver.failBack.mode
phone1
added
reg.n.outboundProxy.failOver.failBack.timeout
phone1
added
reg.n.useCompleteUriForRetrieve
1 sip
added
voIpProt.server.1.failOver.reRegisterOn
sip
added
voIpProt.server.1.failOver.failRegistrationOn
sip
added
voIpProt.server.1.failOver.failBack.mode
sip
added
voIpProt.server.1.failOver.failBack.timeout
sip
added
voIpProt.server.2.failOver.reRegisterOn
sip
added
voIpProt.server.2.failOver.failRegistrationOn
sip
added
voIpProt.server.2.failOver.failBack.mode
sip
added
voIpProt.server.2.failOver.failBack.timeout
sip
added
voipPort.SIP.useCompleteUriForRetrieve
1 sip
added
voIpProt.SIP.outboundProxy.failOver.reRegisterOn
sip
added
voIpProt.SIP.outboundProxy.failOver.failRegistrationOn
sip
added
voIpProt.SIP.outboundProxy.failOver.failBack.mode
sip
added
voIpProt.SIP.outboundProxy.failOver.failBack.timeout
sip
added
voIpProt.H323.blockFacilityOnStartH245
0 sip
added
se.destination
chassis
sip
added
voice.codecPref.IP_5000.G711Mu
2 sip
added
voice.codecPref.IP_5000.G711A
3 sip
added
voice.codecPref.IP_5000.G729AB
4 sip
added
voice.codecPref.IP_5000.G722
1 sip
added
voice.codecPref.IP_5000.iLBC.13_33kbps
sip
added
voice.codecPref.IP_5000.iLBC.15_2kbps
sip
added
voice.gain.rx.analog.chassis.IP_5000
0 sip
added
voice.gain.rx.analog.ringer.IP_5000
0 sip
added
voice.gain.rx.digital.chassis.IP_5000
11
sip
added
voice.gain.rx.digital.ringer.IP_5000
-12
sip
added
voice.gain.tx.analog.chassis.IP_5000
0 sip
added
voice.gain.tx.digital.chassis.IP_5000
15
sip
added
voice.aes.hf.duplexBalance.IP_5000.0
10
sip
added
voice.aes.hf.duplexBalance.IP_5000.1
9 sip
added
voice.aes.hf.duplexBalance.IP_5000.2
8 sip
added
voice.aes.hf.duplexBalance.IP_5000.3
7 sip
added
voice.aes.hf.duplexBalance.IP_5000.4
6 sip
added
voice.aes.hf.duplexBalance.IP_5000.5
5
2.3.4 Configuration File Parameter Changes
Page 12 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
sip
added
voice.aes.hf.duplexBalance.IP_5000.6
4
sip
added
voice.aes.hf.duplexBalance.IP_5000.7
3
sip
added
voice.aes.hf.duplexBalance.IP_5000.8
2 sip
added
voice.ns.hf.IP_5000.enable
1 sip
added
voice.ns.hf.IP_5000.signalAttn
-6
sip
added
voice.ns.hf.IP_5000.silenceAttn
-9
sip
added
voice.rxEq.hf.IP_5000.preFilter.enable
1 sip
added
voice.rxEq.hf.IP_5000.postFilter.enable
0 sip
added
voice.txEq.hf.IP_5000.preFilter.enable
0
sip
added
voice.txEq.hf.IP_5000.postFilter.enable
1
Copyright © 2011 Polycom, Inc. Page 13
Release Notes - SIP Application Changes
2.4 Version 3.2.2
2.4.1 Added or Changed Features
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: VVX1500: Set Default 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: VVX1500: Add support for mutual TLS authentication.
49277: VVX1500: 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.
2.4.2 Removed Features
52522: VVX 1500: Remove “Launchpad” Feature.
2.4.3 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.
Page 14 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application 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.
Copyright © 2011 Polycom, Inc. Page 15
Release Notes - SIP Application Changes
52370: SoundStation IP 7000/HDX Integration: Removing Ethernet cable,
unmutes the Muted phone.
52376: SoundStation IP 6000, 7000: Unable to disable daylight Savings Time.
Introduced in SIP 3.2.0
52381: On some phones; "Retrieve", "Directed" and "Group" soft keys
disappear after entering some digits. This occurs when using the callpark/pick-up feature using SIP signaling. Introduced in SIP 3.2.0.
52415: Enhanced BLF: Ringtones are suppressed when a user is parked
52568: SoundStation IP 7000/HDX Integration Onyx VI: Phone does not play
DTMF tone with default configuration
52580: SoundStation IP 7000/HDX Integration: Delayed DTMF audio feedback
is heard when conferencing third POTS end while using the IP 7000 User
Interface.
52656: VVX 1500: Phone does not support transcoding of video codecs that
are not included in the far-end's capability set
52678: Corporate Directory: When quick/AdvFind search on full last name,
some entries are missing.
52709: License menu reports expiry date of 31-Dec-1969 for license with no
expiry date.
52770: Message-summary SUBSCRIBE is not sent when reg.x.type=shared
52836: Phone allows the user to enter more than maximum allowed (32)
characters in hot dial and contact directory operations. Introduced in SIP 3.2.0.
52860: Split sk should not be available for a Transfer consultation call if the
call per line limit is reached.
52883: In a particular signaling scenario; When a call is placed to a shared
line, the ringer for an IP650 stutters when the call is picked up at another
station.
52943: LLDP reported power usage in logs indicates inappropriate power
consumption.
52950: VQMon: Packet Loss and Burst Gap Loss metrics too high when calling
IVR, caused by valid gap in audio sent from IVR
52963: SoundPoint IP 320, 321, 330, 331: Phone re-boots when user press NN#
from idle screen to invoke Contact Directory entry screen for NN speed dial
index. Occurs if “Presence” feature is enabled. Introduced in SIP 3.2.0.
52971: EFK: Phone re-boots when efkprompt label is longer than 32
characters.
52977: VVX 1500: "Directory" soft key unexpectedly disappears after selecting
"Blind" transfer mode
53007: VVX 1500; VQMon: Phone does not compute RFactor and MOS quality
scores for the G7221C codec
Page 16 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
53034: SUBSCRIBE for BLA with expires:0 received from server is not
recognized as terminating the subscription
53254: VVX 1500: It is not possible to change Auth Password for SIP Lines via
on-phone Admin Settings
53598: Side-tone still present after call hangup on headset. Resets only after
headset button is pressed. Using GN9350e with EHS.
53656: Part number in Phone Status menu is displaying as YYYY-YYYYY-YYY.
Introduced in SIP 3.2.1.
53855: When a phone's extension has an underscore in the name, followed
only by numbers, the underscore is removed in SIP signaling and the device is
not found
53917: Phone Reboots in a certain scenario when using the „Join‟ key
53944: SoundPoint IP 320, 330, 321, 331; SoundStation IP 7000: Phone does
not display Dir soft-key in Korean and Slovenian languages
53946: SoundPoint IP 550, 560, 650, 670: Sometimes the phone displays the
time and date behind a custom idle display.
53975: Phones will not send a SUBSCRIBE message in a certain scenario
when using SCA with barge in enabled.
54034: VVX 1500: Phone generates loud static when CNG packets are
received.
54139: Consultative Transfer uses wrong URI on REFER. Issue introduced in
SIP 3.2.0.
54262: SoundPoint IP 320, 321: Ethernet status menu displays incorrect
information
54631: SoundStation IP7000/HDX Integration: The Voice/Video call type prompt
needs to be removed when hot dialing and pressing the Hook hard key. The
call type should default to Voice by default.
54765: VVX 1500: Phone fails to resend INVITE after 401 from server when
second INVITE is roughly 1500 bytes.
54768: VVX 1500: Phone cannot establish calls properly when booted without
a network connection.
54886: Phone does not send re-Invite with SDP containing session attribute
"a=sendrecv" upon resuming a call when the call is initiated with
"a=sendrecv" offered
54940: New REQUESTS sent directly to far end; route set ignored after a call is
placed on MOH. Loss of audio results.
55052: Additional parameter in From header of INVITE causes 1-way audio
when it is not found in the ACK to a 200 OK
Copyright © 2011 Polycom, Inc. Page 17
Release Notes - SIP Application Changes
File
Change
Attribute
Old value
New value
phone1
added
call.autoOffHook.1.protocol
phone1
added
call.autoOffHook.2.protocol
phone1
added
call.autoOffHook.3.protocol
phone1
added
call.autoOffHook.4.protocol
phone1
added
call.autoOffHook.5.protocol
phone1
added
call.autoOffHook.6.protocol
phone1
added
reg.1.protocol.H323
phone1
added
reg.1.protocol.SIP
phone1
added
reg.1.server.H323.1.address
phone1
added
reg.1.server.H323.1.expires
phone1
added
reg.1.server.H323.1.port
phone1
added
reg.2.protocol.H323
phone1
added
reg.2.protocol.SIP
phone1
added
reg.2.server.H323.1.address
phone1
added
reg.2.server.H323.1.expires
phone1
added
reg.2.server.H323.1.port
phone1
added
reg.3.protocol.H323
phone1
added
reg.3.protocol.SIP
phone1
added
reg.3.server.H323.1.address
phone1
added
reg.3.server.H323.1.expires
phone1
added
reg.3.server.H323.1.port
phone1
added
reg.4.protocol.H323
phone1
added
reg.4.protocol.SIP
phone1
added
reg.4.server.H323.1.address
phone1
added
reg.4.server.H323.1.expires
phone1
added
reg.4.server.H323.1.port
phone1
added
reg.5.protocol.H323
phone1
added
reg.5.protocol.SIP
phone1
added
reg.5.server.H323.1.address
phone1
added
reg.5.server.H323.1.expires
phone1
added
reg.5.server.H323.1.port
phone1
added
reg.6.protocol.H323
phone1
added
reg.6.protocol.SIP
phone1
added
reg.6.server.H323.1.address
phone1
added
reg.6.server.H323.1.expires
phone1
added
reg.6.server.H323.1.port
sip
added
call.autoAnswer.H323
0 sip
added
call.autoAnswer.micMute
1 sip
added
call.autoAnswer.ringClass
4 sip
added
call.autoAnswer.SIP
0 sip
added
call.autoAnswer.videoMute
0 sip
added
call.autoRouting.preference
line
sip
added
call.autoRouting.preferredProtocol
SIP
sip
removedd
httpd.lp.port
sip
removed
httpd.ta.enabled
sip
added
log.level.change.h323
4
2.4.4 Configuration File Parameter Changes
Page 18 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
sip
added
log.level.change.poll
4
sip
added
log.level.change.push
4
sip
added
log.level.change.wmgr
4 sip
removed
mb.launchpad.enabled
sip
removed
mb.main.1.icon
sip
removed
mb.main.1.text
sip
removed
mb.main.1.url
sip
removed
mb.main.2.icon
sip
removed
mb.main.2.text
sip
removed
mb.main.2.url
sip
removed
mb.main.3.icon
sip
removed
mb.main.3.text
sip
removed
mb.main.3.url
sip
removed
mb.main.4.icon
sip
removed
mb.main.4.text
sip
removed
mb.main.4.url
sip
removed
mb.main.5.icon
sip
removed
mb.main.5.text
sip
removed
mb.main.5.url
sip
removed
mb.main.6.icon
sip
removed
mb.main.6.text
sip
removed
mb.main.6.url
sip
added
sec.H235.mediaEncryption.enabled
1 sip
added
sec.H235.mediaEncryption.offer
0 sip
added
sec.H235.mediaEncryption.require
0
sip
added
up.callTypePromptPref
1
sip
added
up.enableCallTypePrompt
1 sip
changed
up.idleBrowser.enabled
0 sip
added
up.manualProtocolRouting
1
sip
added
up.manualProtocolRouting.softKeys
1
sip
changed
video.autoStartVideoTx
1 sip
added
video.callRate
448
sip
added
video.codecPref.H261
4
sip
changed
video.enable
1
sip
added
video.forceRtcpVideoCodecControl
0 sip
changed
video.maxCallRate
512
sip
added
video.profile.H261.annexD
sip
added
video.profile.H261.CifMpi
1
sip
added
video.profile.H261.jitterBufferMax
2000
sip
added
video.profile.H261.jitterBufferMin
150
sip
added
video.profile.H261.jitterBufferShrink
70
sip
added
video.profile.H261.QcifMpi
1
sip
changed
video.screenMode
normal
sip
changed
video.screenModeFS
normal
sip
added
voice.audioProfile.G719.32kbps.payloadTy
pe
107
sip
added
voice.audioProfile.G719.48kbps.payloadTy
pe
108
Copyright © 2011 Polycom, Inc. Page 19
Release Notes - SIP Application Changes
sip
added
voice.audioProfile.G719.64kbps.payloadTy
pe
109
sip
added
voice.audioProfile.G719.jitterBufferMax
200
sip
added
voice.audioProfile.G719.jitterBufferMin
40
sip
added
voice.audioProfile.G719.jitterBufferShrink
1500
sip
added
voice.audioProfile.G719.payloadSize
20
sip
added
voice.codecPref.VVX_1500.G719.32kbps
sip
added
voice.codecPref.VVX_1500.G719.48kbps
sip
added
voice.codecPref.VVX_1500.G719.64kbps
sip
changed
voice.gain.tx.digital.chassis.VVX_1500
6 3 sip
added
voIpProt.H323.autoGateKeeperDiscovery
0
sip
added
voIpProt.H323.dtmfViaSignaling.enabled
1
sip
added
voIpProt.H323.dtmfViaSignaling.H245alpha
numericMode
1
sip
added
voIpProt.H323.dtmfViaSignaling.H245signal
Mode
1
sip
added
voIpProt.H323.enable
0 sip
added
voIpProt.H323.local.port
1720
sip
removed
voIpProt.local.port
sip
added
voIpProt.server.H323.1.address
sip
added
voIpProt.server.H323.1.expires
sip
added
voIpProt.server.H323.1.port
sip
added
voIpProt.SIP.dtmfViaSignaling.rfc2976
sip
added
voIpProt.SIP.enable
1
sip
added
voIpProt.SIP.local.port
5060
File
Change
Attribute
Old
value
New value
Description
sip
added
ind.anim.IP_335.42.frame.1.bitmap
Handset
See Administrator‟s Guide
for SIP 3.2.2 for details
sip
added
ind.anim.IP_335.42.frame.1.duration
1300
See Administrator‟s Guide
for SIP 3.2.2 for details
sip
added
ind.anim.IP_335.42.frame.2.bitmap
PlumHd
See Administrator‟s Guide
for SIP 3.2.2 for details
sip
added
ind.anim.IP_335.42.frame.2.duration
1300
See Administrator‟s Guide
for SIP 3.2.2 for details
2.5 Version 3.2.1 B
2.5.1 Added or Changed Features
48947: Add Support for the SoundPoint IP 335 product.
2.5.2 Removed Features
None.
2.5.3 Corrections
None
2.5.4 Configuration File Parameter Changes
Page 20 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
sip
added
ind.anim.IP_335.43.frame.1.bitmap
Headset
See Administrator‟s Guide
for SIP 3.2.2 for details
sip
added
ind.anim.IP_335.43.frame.1.duration
1300
See Administrator‟s Guide
for SIP 3.2.2 for details
sip
added
ind.anim.IP_335.43.frame.2.bitmap
PlumHd
See Administrator‟s Guide
for SIP 3.2.2 for details
sip
added
ind.anim.IP_335.43.frame.2.duration
1300
See Administrator‟s Guide
for SIP 3.2.2 for details
sip
added
ind.anim.IP_335.44.frame.1.bitmap
Speaker
See Administrator‟s Guide
for SIP 3.2.2 for details
sip
added
ind.anim.IP_335.44.frame.1.duration
1300
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
2.6 Version 3.2.1
2.6.1 Added or Changed Features
None
2.6.2 Removed Features
None.
2.6.3 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
300ms. for HDX integration.
2.6.4 Configuration File Parameter Changes
None.
Copyright © 2011 Polycom, Inc. Page 21
Release Notes - SIP Application Changes
2.7 Version 3.2.0
2.7.1 Added or Changed Features
22527: SoundPoint IP320, 321, 330, 331, 550, 560, 650, 670; SoundStation IP
6000, 7000: Implement „Scrolling Status Bar‟.
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
„Idle Mode‟ consistent with other phone models.
Page 22 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
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 call 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
Enhanced BLF.
Copyright © 2011 Polycom, Inc. Page 23
Release Notes - SIP Application Changes
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
2.7.2 Removed Features
48283: Remove support for SoundPoint IP 301, 501, 600, 601 phones.
48698: Remove support for SoundStation IP 4000
2.7.3 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
line if 1st is unregistered
Page 24 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
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
ID
39949: SoundPoint IP 320, 321, 330, 331; Corporate Directory: Navigation icon
is incorrect when using keypad to navigate
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 a Call display name includes an @ then the display is truncated after
"@" character.
Copyright © 2011 Polycom, Inc. Page 25
Release Notes - SIP Application Changes
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
key is pressed then far end resumes
Page 26 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
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.
Copyright © 2011 Polycom, Inc. Page 27
Release Notes - SIP Application Changes
47578: SoundPoint IP 320, 321, 330, 331; Corporate Directory: The „sticky‟
attributes are not saved.
47612: BLF: Cancelling a Transfer for a call that was initiated using Directed
Call Pick-Up sequence will result in incorrect caller-id display to the user.
47641: SoundStation IP 7000/HDX Integration: Network Link down message
should stay unless phone reboot and comes up with Ethernet cable.
47695: SoundPoint IP 320, 321, 330, 331, 430, 450: When phone has 2
registrations, NewCall soft key is still displayed for alerting call appearance
when there are max call appearances
47699: SoundStation IP6000; XML API; Internal URIs: Tel URI is not working
properly if embedded within a couple of internal URI actions.
47712: SoundPoint IP 320,321,330,331: Local contact directory search does
not always work correctly.
47724: SoundPoint IP 450: Mute icon and Call appearance counter conflict
when DND is turned on and multiple call appearances are present on the
phone
47729: On-hook dialing widget uses multi-tap behavior but is not in multi-tap
mode
47746: NewCall soft key should not be displayed when phone holds max
conference calls
47798: SoundStation IP 7000: Improve location of Transfer and Conference
soft keys during conference setup.
47847: BLF: Monitoring phone stops ringing if shared line is seized while
monitored line has an incoming call
47853: Headset memory mode active: Headset key stops blinking during
incoming call after ending 1st active call.
47862: SoundStation IP6000 : Time and Date doesn't display during call
47863: Phone's HTTP server is sending some HTTP traffic in very small TCP
segments
47916: SoundPoint IP 320, 321, 330, 331: Resume soft key is not available for
2nd call appearance after splitting conf established through Join from
different shared line registrations.
47921: SoundPoint IP 320, 321, 330, 331: The order of call appearances is
different compared with other phones after splitting conf. This discrepancy
results in bringing focus to 2/3 (or 2/4) when split conf.
47929: Rendering special characters like " ' " will break the hyperlink style
display.
47932: Call widget counter (1/n) does not appear while in dial tone state. It
flashes for a fraction of sec and then disappears.
Page 28 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
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
Copyright © 2011 Polycom, Inc. Page 29
Release Notes - SIP Application Changes
48213: XML API; Internal URI: Key:LineX should be executed only if "X" is a
supported line key for that platform.
48333: USB Call Recording: USB busy indicator does not appear on main
screen when recording in progress.
48414: Phone occasionally fails to act on electronic hookswitch up/down
signal from Plantronics and Hydra headsets.
48700: USB call Recording: Stopping Playback through "Back" key not
intuitive
48745: Corporate Directory: LDAP “Critical Extension Error 0x0c” causes CD
Server not responding message from phone.
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
deleting last contact
49572: SoundStation IP7000; Corporate Directory: Numeric characters cannot
be entered in the Quick Search entry field.
49617: Phone does not play dial tone after a hold reminder is played in certain
scenarios.
49619: Call waiting beep does not play on phone when call hold reminder is
set.
Page 30 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
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.
Copyright © 2011 Polycom, Inc. Page 31
Release Notes - SIP Application Changes
49927: SoundPoint IP 320,321,330,331 and VVX 1500: After an AdvFind search,
exit and re-enter Corp Dir menu, phone should displays search bar as
"Search:" not "Search (Filtered):"
49929: SoundStation IP 7000/HDX Integration: SoundStation IP 7000 is not
displaying HDX Extension , when voice call type is set to Auto and phone is
not registered to SIP server
49981: SoundStation IP 7000/HDX Integration: After reboot, 2 digits HDX
extension replaces the last two digits of SIP extension and displays 4
digits(2Digits of sip+2Digits of HDX) with call type HDX.
49982: SoundPoint IP 320, 321, 330, 331: Phone doesn't reconfigure when
DHCP lease expires
49989: SoundStation IP 7000: Phone is adding contact directories from call list
with the existing speed dial number.
49977: SoundPoint IP 320, 321, 330, 331: Phone does not display the selected
status under "MyStat" menu
50090: SoundStation IP 7000: Phone does not display Active Conference
screen on Joining a remotely held SLA call without first holding the local call
50099: Consultative Transfer fails if 2nd leg is forwarding and its 302 response
is handled by proxy
50109: SoundStation IP 7000/HDX Integration: Volume levels are not in Sync
when Dialing a Video call
50110: SoundStation IP 7000/HDX Integration:There is no Enter number
message for Video and audio calls, once the Ethernet is removed.
50115: SoundStation IP 7000/HDX Integration: The DTMF tone of the first digit
is played at HDX volume instead of SoundStation IP 7000 volume.
50118: SoundStation IP 7000/HDX Integration: Dial tone volume and Hands
Free volume are not in Sync.
50137: SoundStation IP 7000/HDX Integration: The volume is reset to default
after the POTS call is connected if voice.volume.persists.handsfree=0
50153: Corporate Directory: Setting the Primary Attribute as „sticky‟
(dir.corp.attribute.1.sticky="1") can give confusing user interface behavior.
50159: Corporate Directory: Quick search on non-null sticky primary filter
missing records
50189: SIP responses missing to-tag after Phone challenges INVITE
50212: Corporate Directory: Scrolling upward for a while, phone displays entry
list not sorted in order
50253: SoundStation IP 7000, Corporate Directory: When edit phone number
attribute in AdvFind menu, pressing on 1/A/a sk creates Encoding sk
50254: Phone does not honor SDP sent in PRACK.
50255: SIP Reliable Provisional responses are not retransmitted.
Page 32 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
50256: When not yet registered, random delay of 30-60 sec between
registration attempts is not observed
50264: Global prefix + not present on calls made from Placed Calls list.
50299: SoundStation IP 7000, Corporate Directory: Quick search text input
starts at the second multitap character instead of the first (e.g. B instead of A
or E instead of D)
50381: SoundPoint IP 320, 321, 330, 331: Pressing left navigation key before
character selection timeout moves cursor 2 spots
50397: SoundStation IP 7000: Phone not displaying licenses correctly in status
screen
50407: Corporate Directory: When server is down with phone connecting to
ldap server, do a quick search, phone displays "No entries found"
50523: SoundPoint IP 320, 321, 330, 331; Corporate Directory: Phone should
display "Contact" title in View menu but it displays quick search bar with a
flashing cursor
50546: When URL dialing disabled, BLIND soft key appears in the 4th soft key
slot, as opposed to the 3rd slot, after pressing TRNSFER.
50811: P-Asserted ID display name should be sticky on UI call appearance and
in placed call list
50869: Phone will only offer SRTP when SRTP crypto suite is selected
50891: SoundStation IP 6000,7000: Resume soft key is not displayed when the
phone is put on hold on another shared line phone.
50989: Receiving a 603 Decline by a BLF monitored user does not play a
reorder tone
51041: X-IdleBrowserSelectUrl: http://url is remembered by the phone even
though idle page doesn't specify it.
51245: BLF state is not updated on receipt of 1st full state NOTIFY after a
reboot
51320: SoundStation IP 7000/HDX Integration: "Conference in Another Video
or phone call?" message is displayed in a loop for each press on "Conf" hard
key.
51432: SoundStation IP 7000/HDX Integration: Conference Hard key Popup
Message need to be altered or displayed appropriately
51554: Phones add an additional CRC to some 802.1X packets received on the
PC port. This causes the 802.1X authentication to fail in some situations.
51567: Server based CFWD/DND sync fails on 3.1.2.0392 [NOTIFY no longer
refreshing target of dialog]
51605: API: Push request will get lost if it follows another push request
immediately.
Copyright © 2011 Polycom, Inc. Page 33
Release Notes - SIP Application Changes
File
Change
Attribute
Old
value
New value
Description
sip
added
call.directedCallPickupMethod
“native” or
“legacy”
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
call.parkedCallRetrieveMethod
“native” or
“legacy”
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
call.parkedCallRetrieveString
Star code
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
added
dialplan.applyToRemoteDialing
0 or 1;
Default is 0
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
2.7.4 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.
Page 34 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
sip
changed
ind.gi.IP_400.4.physX
122
0
modifications
sip
changed
ind.gi.IP_400.5.physX
112
10
sip
changed
ind.gi.IP_4000.6.physH
12
0
sip
changed
ind.gi.IP_4000.6.physW
14
0
sip
changed
ind.gi.IP_4000.6.physX
16 0 sip
changed
ind.gi.IP_4000.6.physY
2 0 sip
changed
ind.gi.IP_450.16.physX
176
196
sip
changed
ind.gi.IP_450.17.physX
176
196
sip
changed
ind.gi.IP_450.18.physX
176
196
sip
changed
ind.gi.IP_450.19.physX
176
196
sip
changed
ind.gi.IP_450.2.physX
40
20
sip
changed
ind.gi.IP_450.3.physH
20 0 sip
changed
ind.gi.IP_450.3.physW
20 0 sip
changed
ind.gi.IP_450.3.physX
20
0
sip
changed
ind.gi.IP_450.3.physY
2
0
sip
changed
ind.gi.IP_600.13.physH
103
111
sip
changed
ind.gi.IP_600.13.physY
0
25
sip
changed
ind.gi.IP_600.4.physY
105
3
sip
changed
ind.gi.IP_600.6.physH
20 0 sip
changed
ind.gi.IP_600.6.physW
20 0 sip
changed
ind.gi.IP_600.6.physX
113
0
sip
changed
ind.gi.IP_600.6.physY
110
0
sip
changed
ind.gi.IP_7000.3.physH
20
0
sip
changed
ind.gi.IP_7000.3.physW
20 0 sip
changed
ind.gi.IP_7000.3.physX
20 0 sip
added
lcl.ml.lang.menu.1.label
简体中文
(zh-cn)
Language selection
displayed in the
appropriate language.
sip
added
lcl.ml.lang.menu.10.label
日本語 (jajp)
sip
added
lcl.ml.lang.menu.11.label
한국어 (kokr)
sip
added
lcl.ml.lang.menu.12.label
Norsk (nono)
sip
added
lcl.ml.lang.menu.13.label
Polski (pl-pl)
sip
added
lcl.ml.lang.menu.14.label
Português
(pt-br)
sip
added
lcl.ml.lang.menu.15.label
сский (ru-
ru)
sip
added
lcl.ml.lang.menu.16.label
Slovenski
(sl-si)
sip
added
lcl.ml.lang.menu.17.label
Español (eses)
sip
added
lcl.ml.lang.menu.18.label
Svenska
(sv-se)
sip
added
lcl.ml.lang.menu.2.label
Dansk (dadk)
sip
added
lcl.ml.lang.menu.3.label
Nederlands
(nl-nl)
sip
added
lcl.ml.lang.menu.4.label
English (enca)
Copyright © 2011 Polycom, Inc. Page 35
Release Notes - SIP Application Changes
sip
added
lcl.ml.lang.menu.5.label
English (engb)
sip
added
lcl.ml.lang.menu.6.label
English (enus)
sip
added
lcl.ml.lang.menu.7.label
Français (frfr)
sip
added
lcl.ml.lang.menu.8.label
Deutsch
(de-de)
sip
added
lcl.ml.lang.menu.9.label
Italiano (it-it)
sip
added
log.level.change.lldp
4
Control the logging detail
level for the LLDP feature.
sip
added
mb.main.autoBackKey
1
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
changed
ramdisk.minfree
3072
3150
Minimum amount of free
space that must be left
after the RAM disk has
been created
sip
changed
se.pat.ringer.13.name
Sample
d 1
Customer ringer file
names.
sip
changed
se.pat.ringer.14.name
Sample
d 2 sip
changed
se.pat.ringer.15.name
Sample
d 3 sip
changed
se.pat.ringer.16.name
Sample
d 4 sip
changed
se.pat.ringer.17.name
Sample
d 5 sip
changed
se.pat.ringer.18.name
Sample
d 6 sip
changed
se.pat.ringer.19.name
Sample
d 7 sip
changed
se.pat.ringer.20.name
Sample
d 8 sip
changed
se.pat.ringer.21.name
Sample
d 9 sip
changed
se.pat.ringer.22.name
Sample
d 10
sip
added
sec.srtp.requireMatchingTag
0 or 1
A flag to determine
whether or not to check
the tag
value in the crypto
attribute in an SDP
answer.
sip
changed
tone.dtmf.rfc2833Payload
101
127
The phone-event payload
encoding in the
dynamic range to be used
in SDP offers.
sip
added
up.idleBrowser.enabled
0 or 1;
default is 0
A flag to determine
whether or not the
background takes priority
over the idle
browser. Used in
conjunction with
up.prioritizeBackground.en
able .
Page 36 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
sip
added
up.prioritizeBackgroundMenuItem.e
nabled
0 or 1;
default is 1.
If set to 1, the “Prioritize
Background” menu is
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
Copyright © 2011 Polycom, Inc. Page 37
Release Notes - SIP Application Changes
sip
added
voIpProt.SDP.early.answerOrOffer
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.
sip
added
voIpProt.SDP.offer.iLBC.13_33kbps
.includeMode
See Administrator‟s Guide
for SIP 3.2.0 for details
sip
changed
voIpProt.server.1.port
5060
port of a SIP
server that accepts
registration
sip
added
voIpProt.server.2.address
sip
added
voIpProt.server.2.expires
Minimum now 10
sip
added
voIpProt.server.2.expires.lineSeize
30
sip
added
voIpProt.server.2.expires.overlap
sip
added
voIpProt.server.2.lcs
sip
added
voIpProt.server.2.port
sip
added
voIpProt.server.2.register
1 sip
added
voIpProt.server.2.retryMaxCount
0 sip
added
voIpProt.server.2.retryTimeOut
0 sip
added
voIpProt.SIP.compliance.RFC3261.
validate.contentLength
If set to 1, validation of the
SIP header
content language is
enabled.
sip
added
voIpProt.SIP.compliance.RFC3261.
validate.uriScheme
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.
phone1
added
attendant.behaviors.display.remote
CallerID.automata
1
Flags to determine
whether or
not remote party caller ID
information is presented to
the
attendant.
phone1
added
attendant.behaviors.display.remote
CallerID.normal
1 phone1
added
attendant.behaviors.display.spontan
eousCallAppearances.automata
0
Flags to determine
whether or
Page 38 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
phone1
added
attendant.behaviors.display.spontan
eousCallAppearances.normal
1
not a call appearance is
spontaneously presented
to the
attendant when calls are
alerting
on a monitored resource
phone1
added
attendant.resourceList.x.address
The value of
x depends
on the
phone
For IP 450:
x=1-2; IP
550, IP 560:
X=1-3; IP
650, IP 670:
x=1-47
The user referenced by
attendant.reg="" will
subscribe to this
URI for dialog.
phone1
added
attendant.resourceList.x.label
Text label to appear on the
display
adjacent to the associated
line key
phone1
added
attendant.resourceList.x.type
“normal”
Type of resource being
monitored.
phone1
changed
attendant.ringType
1
phone1
added
dialplan.1.applyToTelUriDial
1
When present, and if
dialplan.x.digitmap is not
Null, this attribute
overrides the
global dial plan defined in
the
sip.cfg configuration file.
phone1
added
dialplan.2.applyToTelUriDial
1 phone1
added
dialplan.3.applyToTelUriDial
1
phone1
added
dialplan.4.applyToTelUriDial
1
phone1
added
dialplan.5.applyToTelUriDial
1 phone1
added
dialplan.6.applyToTelUriDial
1 phone1
changed
divert.noanswer.1.timeout
60
55
Modified No Answer
Timeout
phone1
changed
divert.noanswer.2.timeout
60
55
phone1
changed
divert.noanswer.3.timeout
60
55
phone1
changed
divert.noanswer.4.timeout
60
55
phone1
changed
divert.noanswer.5.timeout
60
55
phone1
changed
divert.noanswer.6.timeout
60
55
phone1
added
reg.1.server.2.address
See Administrator‟s Guide
for SIP 3.2.0 for details
phone1
added
reg.1.server.2.expires
phone1
added
reg.1.server.2.expires.lineSeize
phone1
added
reg.1.server.2.expires.overlap
phone1
added
reg.1.server.2.lcs
phone1
added
reg.1.server.2.port
phone1
added
reg.1.server.2.register
phone1
added
reg.1.server.2.retryMaxCount
phone1
added
reg.1.server.2.retryTimeOut
phone1
added
reg.2.musicOnHold.uri
phone1
added
reg.2.server.1.lcs
phone1
added
reg.2.server.2.address
phone1
added
reg.2.server.2.expires
phone1
added
reg.2.server.2.expires.lineSeize
phone1
added
reg.2.server.2.expires.overlap
phone1
added
reg.2.server.2.lcs
phone1
added
reg.2.server.2.port
Copyright © 2011 Polycom, Inc. Page 39
Release Notes - SIP Application Changes
phone1
added
reg.2.server.2.register
phone1
added
reg.2.server.2.retryMaxCount
phone1
added
reg.2.server.2.retryTimeOut
phone1
added
reg.2.tcpFastFailover
phone1
added
reg.3.musicOnHold.uri
phone1
added
reg.3.server.1.lcs
phone1
added
reg.3.server.2.address
phone1
added
reg.3.server.2.expires
phone1
added
reg.3.server.2.expires.lineSeize
phone1
added
reg.3.server.2.expires.overlap
phone1
added
reg.3.server.2.lcs
phone1
added
reg.3.server.2.port
phone1
added
reg.3.server.2.register
phone1
added
reg.3.server.2.retryMaxCount
phone1
added
reg.3.server.2.retryTimeOut
phone1
added
reg.3.tcpFastFailover
phone1
added
reg.4.musicOnHold.uri
phone1
added
reg.4.server.1.lcs
phone1
added
reg.4.server.2.address
phone1
added
reg.4.server.2.expires
phone1
added
reg.4.server.2.expires.lineSeize
phone1
added
reg.4.server.2.expires.overlap
phone1
added
reg.4.server.2.lcs
phone1
added
reg.4.server.2.port
phone1
added
reg.4.server.2.register
phone1
added
reg.4.server.2.retryMaxCount
phone1
added
reg.4.server.2.retryTimeOut
phone1
added
reg.4.tcpFastFailover
phone1
added
reg.5.musicOnHold.uri
phone1
added
reg.5.server.1.lcs
phone1
added
reg.5.server.2.address
phone1
added
reg.5.server.2.expires
phone1
added
reg.5.server.2.expires.lineSeize
phone1
added
reg.5.server.2.expires.overlap
phone1
added
reg.5.server.2.lcs
phone1
added
reg.5.server.2.port
phone1
added
reg.5.server.2.register
phone1
added
reg.5.server.2.retryMaxCount
phone1
added
reg.5.server.2.retryTimeOut
phone1
added
reg.5.tcpFastFailover
phone1
added
reg.6.musicOnHold.uri
phone1
added
reg.6.server.1.lcs
phone1
added
reg.6.server.2.address
phone1
added
reg.6.server.2.expires
phone1
added
reg.6.server.2.expires.lineSeize
phone1
added
reg.6.server.2.expires.overlap
phone1
added
reg.6.server.2.lcs
phone1
added
reg.6.server.2.port
phone1
added
reg.6.server.2.register
Page 40 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
phone1
added
reg.6.server.2.retryMaxCount
phone1
added
reg.6.server.2.retryTimeOut
phone1
added
reg.6.tcpFastFailover
2.8 Version 3.1.6
2.8.1 Added or Changed Features
None.
2.8.2 Removed Features
None.
2.8.3 Corrections
54423: SoundPoint IP 601: Phone reboots under heavy SIP traffic while using
Buddy Watch as a BLF.
54479: SoundPoint IP 601 + 32 member BLF: After upgrading from 2.1.2 to
3.1.3RevB, users experience a delay in transferring calls using the Transfer
key.
2.8.4 Configuration File Parameter Changes
None.
2.9 Version 3.1.5 (Limited Distribution)
2.9.1 Added or Changed Features
None.
2.9.2 Removed Features
None.
2.9.3 Corrections
54165: Phone cannot pick up call off hold after it receives NOTIFY with dialog
state="full" in response to its BLA re-subscribe
2.9.4 Configuration File Parameter Changes
None.
Copyright © 2011 Polycom, Inc. Page 41
Release Notes - SIP Application Changes
2.10 Version 3.1.4
2.10.1 Added or Changed Features
None
2.10.2 Removed Features
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.
2.10.3 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
2.10.1 Configuration File Parameter Changes
None.
2.11 Version 3.1.3 C
2.11.1 Added or Changed Features
Add Support for the SoundPoint IP 321 and 331 products.
2.11.2 Removed Features
None.
2.11.3 Corrections
None.
Page 42 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
2.11.4 Configuration File Parameter Changes
None.
2.12 Version 3.1.3 B
2.12.1 Added or Changed Features
None.
2.12.2 Removed Features
None.
2.12.3 Corrections
50103 SoundStation IP 7000/HDX: Volume change before dialing is discarded
after the POTS call is established
50104: Corporate Directory: If ViewPersistency is enabled, Scrolling down the
list of results from an Advanced Find querey, after exit ->re-enter->scroll up,
attribute filter in previous AdvFind is not maintained
50117: SoundStation IP 7000/HDX: Incoming POTS call resets the Ringer
volume.
2.12.4 Configuration File Parameter Changes
None.
2.13 Version 3.1.3 (Limited Release – Version 3.1.3.0336 )
2.13.1 Added or Changed Features
45869: Corporate Directory: Add support for LDAP directory queries using
VLV Indexing.
47179: Extend fast-fail over mechanism to transactions initiated over TCP
transport
47493: Corporate Directory: Improvements to User Interface. See Technical
Bulletin TB 41137 for details.
47495: Corporate Directory: Screen Idle Timeout needs to be reset whilst a
Corporate Directory search is in process
48183: VVX 1500: Add network jitter computation and reporting for video
packet channels
48467: VVX 1500: Touching the LCD screen at any location should wake the
LCD from the "dim" state to full brightness.
48484: IP7000/HDX: Allow Configuration control of the Dialtone sound level
when adding a POTS call to an existing Video call.
Copyright © 2011 Polycom, Inc. Page 43
Release Notes - SIP Application Changes
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
2.13.2 Removed Features
47376: Remove License Requirement on uaCSTA feature
2.13.3 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
making a video call
Page 44 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
48039: BLF: Phone plays the „Attendant Ring-Tone‟ instead of the „Regular
Ring-Tone‟ if the remote line and local phone are both „Ringing‟ and the
remote line is answered and then put on Hold.
48046: On G.729ab gateway calls speaker phone volume is not loud enough
for low level signals
48076: BLF: Attendant phone does not automatically get placed on Hold if a
BLF or speed dial key is used to dial whilst an active call is in process on the
attendant phone. Only occurs if call.stickyAutoLineSeize=”1”.
48123: SoundStation IP 4000/6000/7000: Clock time does not increment while a
call is active if the idle browser is enabled.
48171: De-registration attempts do not authenticate and so fail to de-register
some lines.
48280: SoundStation IP 6000, 7000: When using TFTP or FTP as the
provisioning Server Type, phone does not save directory entries locally when
TFTP or FTP server is not available.
48385: VVX 1500: SSRC header field is not correct for RFC2833 packets.
48462: SoundPoint IP 501: Ring LED indicator continues flashing even when
the call is answered if an INVITE with “sendonly” SDP is received by the
phone.
48485: VVX 1500: Audio call recording during video calls may fail with certain
USB drives.
48577: SoundPoint IP 430: Default headset gains not correctly set which may
result in poor audio quality with certain headsets.
48591: VVX 1500: Click-to-Hold does not work correctly.
48605: call.stickyAutoLineSeize is not applied correctly when a line is ringing
and SilentRing is selected
48615: If call.StickyAutoLineSeize=”1”: Transfer fails if attempted whilst a
second call is alerting.
48667: If there is an incoming call while there is an existing outgoing call in the
proceeding state, the phone will not audibly alert the user for the incoming call
48668: 401 Authentication challenge to a VQMon PUBLISH may cause the
phone to reboot.
48672: Received volume on the handset is lower than desired for low input
signal levels. Addressed by adding 4dB gain at low input levels on the
handset. Gain at high input levels is unchanged.
48685 In SIP 3.1.2 the MWI NOTIFY must have the message summary for the
MWI LED to be lit.
48697: An incoming call without Caller ID Name but with Caller ID Number is
matched with the first local contact that has Name blank.
48699: TelURI doesn't process "tel://*50"
Copyright © 2011 Polycom, Inc. Page 45
Release Notes - SIP Application Changes
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: VVX1500 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
line has 1 or multiple incoming calls being ended
Page 46 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
49810: VVX 1500 seizes line key 1 when "call.stickyAutoLineSeize=1" and the
speed dial key is used to dial.
Copyright © 2011 Polycom, Inc. Page 47
Release Notes - SIP Application Changes
.cfg File
Action
Parameter
Description
sip
added
voIpProt.SIP.serverFeatureControl.localPr
ocessing.dnd
If set to 0 and
voIpProt.SIP.serverFeatureContr
ol.dnd ="1", the phone will not perform
local DND call behavior.
If set to 1 or Null, the phone will perform
local DND call behavior on all calls
received.
sip
added
voIpProt.SIP.serverFeatureControl.localPr
ocessing.cf
If set to 0 and
voIpProt.SIP.serverFeatureContr
ol.cf="1", the phone will not perform
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.
2.13.4 Configuration File Parameter Changes
Page 48 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
sip
added
dir.corp.vlv.sortOrder
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
2.14 Version 3.1.2 B
2.14.1 Added or Changed Features
Add Support for the VVX 1500 product.
2.14.2 Removed Features
None.
2.14.3 Corrections
None.
2.14.4 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.
2.15 Version 3.1.2
2.15.1 Added or Changed Features
34787: Add Support for ACD Call Center Agent functionality using the „Feature
Copyright © 2011 Polycom, Inc. Page 49
Synchronization‟ method. See Technical Bulletin 34787 for details.
Release Notes - SIP Application Changes
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.
2.15.2 Removed Features
N/A
2.15.3 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
call.stickyAutoLineSeize.onHookDialing = "0"
Page 50 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
45102: SoundStation IP 7000: 1/A/a soft key is missing in Corp Dir search
screen.
45169: When using sampled audio as local hold notification Local hold
notification may play inaudibly or muffled.
45273: SoundStation IP4000 will not register when qos.ip.callControl.dscp =
"24"
45422: Adding speed dial entry using Expansion Module may place new entry
in an unexpected place
45479: SoundStation IP7000: Time&Date setting returns to the default when
the phone is rebooted.
45715: Ringing stops when users goes on-hook after lifting handset during
incoming call when up.offHookAction.none = 1
45799: XML API: Internal URIs: softkey:Exit, softkey:Submit and softkey:Reset
do not work when called from hyperlink anchor tags
46051: Manage N-way conference menu has overlapping items if long caller-
ids are present.
46144: JPEG decoder fails on some files
46242: XML API: If an account supports 2 line keys, API notifications of call
events are sent for only 1 of them
46293: Phones may lock up if a CHECK-SYNC is received while a CHECK-
SYNC is in progress
46422: Five to six second delay in UI when using the SPLIT softkey to cancel a
transfer
46488: Phone plays continuous Reorder tone if a BLA line is successfully
seized with a new line ID after a previous GLARE response.
46539: Centralized Conferencing: Conference call is terminated if the phone
tries to join a conference that has reached its maximum number of
participants.
46553: When call.stickyAutoLineSeize=”1”, an active call is not put on hold
when 2nd call is made via speed dial or from calls list menu
46569: No ACK sent after receiving VM 200 OK w/ SDP, CANCEL sent 60 secs
later.
46610: Errors in Polish language dictionary
46737: BLF: Softkeys & Call appearance disappears on the console phone in a
certain scenario using a shared line.
46757: XML API: Issue with order of call appearances on a single line
registration and single line key
46763: XML API: URI softkey:exit does not work when executed from softkey
or hyperlink anchor XHTML tags
Copyright © 2011 Polycom, Inc. Page 51
Release Notes - SIP Application Changes
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.
Page 52 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
47531: Using a BLF or Speed Dial key for a Conference operation does not
work.
47537: If call.stickyAutoLineSeize=”1”, initiating a second call whilst a first call
is in the “Outgoing Proceeding” State will result in two calls in the Proceeding
state
47681: BLF: Attendant may not be able to perform directed call pick up on a
monitored line if using a shared line.
47705: When a phone holds a call, press headset button->EndCall sk->NewCall
sk, the phone does not switch back to hands free mode
47716: Config call.stickyAutoLineSeize="1", phone does not seize correct line
key when dialing from Call List or Contact Directory
47728: SoundPoint IP 601: Attendant does not display incoming call
appearance and does not hear attendant ringing tone when a monitored line is
on the 2nd or 3rd Expansion Module
47741: When using 1, 3, 7, 5 key combo to reset flash settings, the UI has
some errors.
47866: SoundPoint IP 320/330/430/450/550/560/650/670: The phone may reboot
when hold reminder tone is enabled and a call is active on the speakerphone.
47537: If call.stickyAutoLineSeize=”1”, initiating a second call whilst a first call
is in the “Outgoing Proceeding” State will result in two calls in the Proceeding
state
47538: On-hook entered digits on a BLF attendant phone are erased if a
remote BLF phone in ringing state is answered on the remote BLF phone.
47559: In some scenarios a BLF attendant phone incorrectly plays the
attendant ringing tone.
Copyright © 2011 Polycom, Inc. Page 53
Release Notes - SIP Application Changes
.cfg File
Action
Parameter
Description
phone1
added
acd.reg
See Technical Bulletin34787 for
details
phone1
added
acd.stateAtSignIn
sip
added
voIpProt.SIP.acd.signalingMethod
sip
added
voIpProt.SIP.compliance.RFC3261.validat
e.contentLanguage
If set to 1, validation of the SIP header
content language is enabled.
If set to 0 or Null, validation is disabled.
sip
removed
bg.gray.selection
Modified the method in which the
background settings are managed
across multiple phone models
sip
added
bg.hiRes.gray.selection
sip
removed
bg.color.selection
sip
added
bg.hiRes.color.selection
sip
added
bg.medRes.gray.selection
sip
changed
ind.gi.IP_600.13.physH
Changed from 109 to 103
sip
changed
ind.gi.IP_7000.7.physH
Changed from 60 to 76
sip
added
log.level.change.cmr
Control the logging detail level for
individual components: call media
recording, call media playback, USB I/O
respectively.
sip
added
log.level.change.cmp
sip
added
log.level.change.usbio
sip
added
prov.quickSetup.enabled
See Technical Bulletin 45460 for
details
sip
added
pnet.hdx.ext
HDX Extension Number. For HDX/IP
7000 integration
2.15.4 Configuration File Parameter Changes
2.16 Version 3.1.1 B
2.16.1 Added or Changed Features
None.
2.16.2 Removed Features
None.
2.16.3 Corrections
47034: SoundStation IP 7000 connected to HDX: Cannot make POTS call when
Ethernet is connected and Call preference configured to Auto.
47082: SoundStation IP 7000 connected to HDX: Phone does not Mute on
Auto-Answer.
47251: SoundStation IP 7000 connected to HDX: When participants in a multi-
point call are disconnected the phone unmutes the local phone incorrectly.
47432: SoundStation IP 7000 connected to HDX: In a certain scenario the
phone sends audio to the far end even though it shows that the call is muted.
Page 54 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
2.16.4 Configuration File Parameter Changes
2.17 Version 3.1.1
2.17.1 Added or Changed Features
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
2.17.2 Removed Features
N/A
2.17.3 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
Copyright © 2011 Polycom, Inc. Page 55
Release Notes - SIP Application Changes
45039: Unnecessary Refer is sent by phone as it is being blind transferred to a
conference focus
45073: Phones do renew their DHCP Lease when they have been operational
for longer than 99 days.
45187: Voice streams are not resumed automatically after a play uri
45316: Phones can re-boot when a they are sent a check-sync while under
some load
45364: In a certain scenario, when SCA phone views remote shared line's call
appearance list, the UI does not return back to its previous state
45380: XML API: Phone may reboot when accessing XHTML pages containing
<softkey> tag
45386: When remote shared line is on hold, press NewCall >Cancel/EndCall sk,
both shared line displays hold screen
45410: Phone‟s micro-browser is not honoring DNS TTL.
45657: BLF Console Phone does not behave correctly when List URI is
removed from the server configuration
45750: Rapidly pressing a new speed dial key after it has just been entered
may cause the phone to re-boot
45602: Early dialog state not reported by NOTIFY if the far end does not
support (100rel) or send PRACK
45713: dialog-info document is empty in NOTIFY to subscription 2,3,,,n when
dialog state is terminated
45827: Entered number cannot be edited by pressing left arrow key to move
cursor to the left in some scenarios
45870: When bitmap is loaded as background for idle display and either the
plus or minus volume key is pressed, the volume indicator graphic does not
clear automatically
45895: Phone will not dial from contact directory when separators are part of
the contact e.g. 604-450-1234
45954: SUBSCRIBE to phone with expires less than 2 seconds will never
receive a NOTIFY
46047: BLF lamps remain on when no explicit "terminated" state sent for BLF
but it has been omitted in the "Full" list
46407: Soft keys do not show up after a call is taken off hold quickly - one-way
audio issue
46412: BLF: Memory Fragmentation and leak with receipt of BLF messaging
46500: BLF: DisplayName is not included in Remote Identity of Dialog when
phone receives REQUEST
Page 56 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
.cfg File
Action
Parameter
Description
sip
changed
voice.gain.rx.digital.chassis.IP_330
voice.gain.rx.digital.chassis.IP_430
voice.gain.rx.digital.chassis.IP_650
voice.gain.rx.digital.chassis.IP_7000
voice.gain.rx.digital.chassis.IP_6000
voice.gain.rx.digital.chassis.IP_450
Changed from 6 to 5
46543: BLA: phone should NOT send dialog NOTIFY with terminated after
receiving a cancel
46486: Enabling Idle Browser on IP330 may cause dialed digits to not display
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.
2.17.4 Configuration File Parameter Changes
2.18 Version 3.1.0 C
2.18.1 Added or Changed Features
Add Support for the SoundPoint IP 450 product.
2.18.2 Removed Features
None.
2.18.3 Corrections
None.
Copyright © 2011 Polycom, Inc. Page 57
Release Notes - SIP Application Changes
.cfg File
Action
Parameter
Description
sip
added
voice.gain.rx.analog.chassis.IP_450
voice.gain.rx.analog.ringer.IP_450
voice.gain.rx.digital.chassis.IP_450
voice.gain.rx.digital.ringer.IP_450
voice.gain.tx.analog.chassis.IP_450
voice.gain.tx.digital.handset.IP_450
voice.gain.tx.digital.headset.IP_450
voice.gain.tx.digital.chassis.IP_450
voice.rxEq.hs.IP_450.preFilter.enable
voice.rxEq.hs.IP_450.postFilter.enable
voice.rxEq.hd.IP_450.preFilter.enable
voice.rxEq.hd.IP_450.postFilter.enable
voice.rxEq.hf.IP_450.preFilter.enable
voice.rxEq.hf.IP_450.postFilter.enable
voice.txEq.hs.IP_450.preFilter.enable
voice.txEq.hs.IP_450.postFilter.enable
voice.txEq.hd.IP_450.preFilter.enable
voice.txEq.hd.IP_450.postFilter.enable
voice.txEq.hf.IP_450.preFilter.enable
voice.txEq.hf.IP_450.postFilter.enable
voice.handset.rxag.adjust.IP_450
voice.handset.txag.adjust.IP_450
voice.handset.sidetone.adjust.IP_450
voice.headset.rxag.adjust.IP_450
voice.headset.txag.adjust.IP_450
voice.headset.sidetone.adjust.IP_450
Add DSP parameters for IP 450
platform.
sip
added
bitmap.IP_450.*
ind.anim.IP_450.*
ind.gi.IP_450.*
Add UI parameters for IP 450
platform.
2.18.4 Configuration File Parameter Changes
2.19 Version 3.1.0 B
2.19.1 Added or Changed Features
None.
2.19.2 Removed Features
None.
2.19.3 Corrections
45605: Missing closing XML tag in a configuration file causes a phone reboot
2.19.4 Configuration File Parameter Changes
None.
Page 58 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
2.20 Version 3.1.0 (Limited Distribution; build-id 3.1.0.0073)
This version should be replaced by 3.1.0RevB
2.20.1 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
the user
Copyright © 2011 Polycom, Inc. Page 59
Release Notes - SIP Application Changes
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.
Page 60 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
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.
2.20.2 Removed Features
N/A
2.20.3 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
license key expiration, post license.polling.time
37233: SoundPoint IP330, IP430, IP650, IP550 and IP4000 phones malfunction
if you enter > 40 digit contact number in directory.xml file.
37449: The phone may re-boot when the user tries to end a local conference if
the call server does not respond to the REFER message.
37580: DoS: Multicast rate limiting is not enabled on IP601
Copyright © 2011 Polycom, Inc. Page 61
Release Notes - SIP Application Changes
37848: LED indication functionality is not consistent among platforms when
IMs are exchanged between phones while on "Instant messages" screen.
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.
Page 62 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
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
39987: Corporate Directory: In phone CD status menu the port displayed is
wrong, though internally the functionality is fine.
39988: DNS NAPTR mis-configuration can cause phone to reset
39996: Only one of the two calls appears on the UI when transferring a
conference between shared lines
40005: Phone does not remove BLFs from the U/I if all monitored users are
removed at once.
40057: Volume Control not visible when adjusting volume while in Manage
Conference menu
40066: N-way conf: In manage menu, Animations icon disappear from the
screen when user selects the participant by pressing its corresponding
number (digit) on dial pad.
40101: USB: Backlight does not get turned on when USB memory stick is
attached/removed.
40117: Corporate Directory: Modify algorithms for displaying CD status and
entry details.
40125: CMR/P: In Browse Files menu the file name gets appended with ellipses
(...) when exit from the Delete screen.
40126: CMR/P: File name is partially truncated at the beginning in audio player
screen in a certain scenario.
40197: CMR/P: The menu title for "Browse Files..." option is "USB Device"
which is duplicate of parent menu screen.
40328: Phone hanging on HTTP PUT with authentication
40399: Phones generates multiple SOA queries and eventually locks up if the
DNS domain is incorrectly configured.
40400: Phone issuing DHCP Inform packet when it doesn't need to.
40416: Backlight does not go to Dim mode (medium) under these scenarios
(when On intensity=HIgh, Idle intensity = Medium)
40436: Backlight intensity should not change from medium to low under these
scenarios when configured (On=medium & Idle = Off).
40445: Place an incoming call to a phone that enables call forward, screen
flickers incoming caller id for 1 time if the phone is in dial tone state
40503: [Corporate Directory] The scroll down bar is still available even if
corporate directory list is accessed to the end.
40561: [Presence] Backspace or "<<" softkey is not available on Add Buddy
Page for IP 4000 and IP 6000 phones.
Copyright © 2011 Polycom, Inc. Page 63
Release Notes - SIP Application Changes
40562: [Presence] The first option in the "Mystat" list gets highlighted even if
option other than the first option is selected.
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 sk.
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.
Page 64 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
42841: Echo is experienced when calling IP 650 to IP 650 using G.722 HD at full
volume.
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
disabled.
Copyright © 2011 Polycom, Inc. Page 65
Release Notes - SIP Application Changes
.cfg File
Action
Parameter
Description
sip
added
voIpProt.SIP.strictLineSeize
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
sip
changed
voice.aec.hs.enable
The default value is enabled (1).
sip
changed
voice.aes.hs.enable
The default value is enabled (1).
sip
added
call.directedCallPickupString
The star code to initiate a directed call
pickup.
2.20.4 Configuration File Parameter Changes
Page 66 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
sip
added
dir.corp.pageSize
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.
sip
added
apps.push.username
The user name to access the push
server URL.
sip
added
apps.push.password
The password to access the push
server URL.
Copyright © 2011 Polycom, Inc. Page 67
Release Notes - SIP Application Changes
sip
added
softkey.x.label
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.
Page 68 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
sip
added
softkey.x.use.hold
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.
Copyright © 2011 Polycom, Inc. Page 69
Release Notes - SIP Application Changes
sip
added
softkey.feature.mystatus
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.
2.21 Version 3.0.4
Note that Versio 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.
2.21.1 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.
2.21.2 Removed Features
N/A
2.21.3 Corrections
43264: Phone is not able to answer calls due to duplicate INVITEs with same
details and new BRANCH ID
Page 70 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
.cfg File
Action
Parameter
Description
sip
changed
voice.aec.hs.enable
voice.aes.hs.enable
voice.ns.hs.enable
Changed default value from „0‟ to „1‟
sip
changed
voice.gain.rx.digital.chassis.IP_330
voice.gain.rx.digital.chassis.IP_430
voice.gain.rx.digital.chassis.IP_650
Changed default value from „6‟ to „5‟
43513: SoundPoint IP 650 to 650 calls experiencing Echo at full volume on the
handset
44029: When ANALOG HEADSET MODE is set to JABRA, there is no audio call
waiting tone
44066: Ringer (including call waiting tone) diminishes on some phones over
time and stops being audible
44413: Speed dial labels on line leys are labeled 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 causing URL
Call Disabled due to unnecessary attempt to provision CFS license file via
HTTPS
44520: Phone generating an unexpected NOTIFY on incoming call, putting BLA
status out of sync
44763: Phones ignoring DNS SRV records response from Session Border
Controller in certain scenario
44818: Danish dictionary is Chinese
45073: Phones do not renew their DHCP Lease when they have been
operational for longer than 99 days.
45118: Digest Authentication for SIP transactions fail when "Digest" token is
all lower-case
45221: Oneway voice in handset/headset mode during call waiting when
call.callWaiting.ring = ring is set.
45719: Corporate Directory: Phone not sending correct details when
connecting to SUNldap Server
45761: DND Sync feature failing across reSUBSCRIBE
2.21.4 Configuration File Parameter Changes
2.22 Version 3.0.3 B
Change made applies to the SoundStation IP 7000 product only.
2.22.1 Added or Changed Features
None.
2.22.2 Removed Features
None.
Copyright © 2011 Polycom, Inc. Page 71
Release Notes - SIP Application Changes
2.22.3 Corrections
41974: SoundStation IP 7000 occasionally reboots when the idle browser is
enabled
2.22.4 Configuration File Parameter Changes
None.
2.23 Version 3.0.3
2.23.1 Added or Changed Features
39423: Change default boot config and packaged sip.cfg value for
parameter voice.vad.signalAnnexB
40385: Add config parameters voIpProt.SIP.strictLineSeize,
reg.x.strictLineSeize and voIpProt.SIP.lineSeize.retries
40387: SIP stack will use config parameter
voIpProt.SIP.strictLineSeize and voIpProt.SIP.lineSeize.retries to make
fault-tolerant behavior optional
40447: Add a User Option to Restart the phone
2.23.2 Removed Features
None
2.23.3 Corrections
39635: Phones configured for a bridged line appearance reboot when they
receive an improperly forked duplicate packet.
39792: The phone is requesting a SIP URI on transfer instead of a number with
some call servers.
40175: Digitmap problem with IP330 and IP320s not processing single digit
map entry correctly
40287: Phone is not returning fast busy on a timeout when sending "TRYING"
state; it continues to send call "EARLY" causing BLA sync issues
40318: Buddy Status indicator not working when a function key is mapped to a
speed dial
40632: Phones hang at the welcome screen when DHCP server specifies a
subnet mask of 255.255.254.0
40673: Phone does not handle NOTIFY message correctly in Glare (race
condition)
40709: Phone responding to subscribe that does not match its configuration
40766: Phone must match To: header with third-party subscribe
41203: Phones not responding to DHCP offer using an option other than 160 if
Option 160 also has an entry. Affects SoundPoint IP 320, 330, 430, 550, 560, 650
phones.
Page 72 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
41351: Call lists may show SIP URI on SoundPoint IP 330/320 phones.
41403: CMR/P: Wrong popup appears when usb is removed after exiting from
the playback to the browse files menu
41475: After upgrade to SIP 3.0 The SIP Config option
msg.bypassInstantMessage=1 does not work correctly.
41614: Phone repeating USER AGENT string in HTTP request.
41645: Transfer of Held call causes party on Hold to automatically resume in
certain call server interactions.
41654: CMR/P: Call gets answered in speaker mode when off-hook if an
incoming call happens while in audio player screen.
41657: CMR/P: Headset memory persistence status goes wrong if an incoming
call happens while in audio player screen.
41666: CMR/P: While in audio player screen, ringing for an incoming call
happens in wrong termination mode. It should always happen on speaker.
41789: AsFeature doesn't reSUBSCRIBE after losing the TLS connection
41808: Idle logo does not display correctly in certain configurations.
41903: Corporate Directory searches may not return complete results if results
contain Unicode character values > 127 (server supports sorting)
41926: Navigation select button does not get call details.
41983: SCA Caller ID displays wrong direction as "From:" when remote shared
line places an outgoing call
42605: Speed dial shortcut should not apply if contact directory is disabled on
SoundPoint IP 330/320 phones
Copyright © 2011 Polycom, Inc. Page 73
Release Notes - SIP Application Changes
.cfg File
Action
Parameter
Description
sip
added
voIpProt.SIP.strictUserValidation
If set to “1”, forces phone to match
user portion of signaling exactly. If set
to “0”, phone will use first registration
if the user part does not match any
registration
sip
added
voIpProt.SIP.strictLineSeize
If set to “1”, forces phone to wait for
200 OK when receiving a TRYING
notify.
sip
added
voIPProt.SIP.lineSeize.retries
Controls the number of times the
phone will retry a notify when
attempting to seize a line (BLA). Valid
values are 3 to 10. Note that in this
release, a value of 3 results in 10. A
value of 2 can be used to get 3
retries.
phone1
added
reg.n.strictLineSeize
If set to “1”, forces phone to wait for
200 OK on registration n when
receiving a TRYING notify.
If this parameter is Null,
voIpProt.SIP.strictLineSeize is checked.
This parameter takes precedence.
2.23.4 Configuration File Parameter Changes
2.24 Version 3.0.2 C
2.24.1 Added or Changed Features
None.
2.24.2 Removed Features
None.
2.24.3 Corrections
42034: Phone freezes when booting from TFTP server in certain scenarios
42060: When an IP601 with Expansion Modules attached is configured with many
speed-dials with long names. Removing or adding a speed-dial during a period of
high activity (e.g. call in progress) may result in sluggish UI response or in
extreme cases re-boot.
2.24.4 Configuration File Parameter Changes
None.
2.25 Version 3.0.2 B (Limited Release – build-id 3.0.2.0917)
2.25.1 Added or Changed Features
Page 74 Copyright © 2011 Polycom, Inc.
Add Support for the SoundPoint IP 670 product
Release Notes - SIP Application Changes
Add Support for the SoundStation IP 6000 product.
Add Support for the SoundStation IP 7000 product.
39292: Add dynamic test for un-recognized USB devices.
39532: After 500 Glare response, phone should retry call attempt on a
different line ID
39585: Add support for JPEG images (in addition to BMP format)
40351: Add additional USB flash drives to the internal list of supported
drives
40591: Add background preference configuration to the phone‟s
configuration web server
41025: Set default LDAP Corporate Directory background re-sync period
to 24 hours
41045: Make initial background LDAP Contact Directory synchronization
optional
41363: Add additional graphic backgrounds to the IP 550, 560, 650
phones.
41517: Add JPEG support to the micro-browser
2.25.2 Removed Features
None.
2.25.3 Corrections
38539: Micro-Browser does not display Asian fonts on IP 550, 560 and 650
phones.
39603: Rapid hold-resume with SRTP can cause one-way audio
39608: Phone does not play ring tone when conference put on hold in certain
scenarios.
39610: Idle display not fully cleared when making new call.
39657: Phone may reboot if user removes USB flash drive while recording is in
progress
39678: Authorization response changes during multi-stage dialing
39716: Speed dial from up arrow shortcut using speed dial index does not work
correctly when BLF is configured
39932: Unicode text entry does not work correctly.
39979: SoundPoint IP 301, 501, 601 phones with SRTP disabled reject calls
offering both SRTP and non-SRTP media
40115: CMR/P: File browser continues to display file in file list after user has
deleted file
Copyright © 2011 Polycom, Inc. Page 75
Release Notes - SIP Application Changes
40266: Voice Quality Metrics incorrectly reports packet losses when VAD is
enabled
40346: Corporate Directory: Improve message when connection is lost after CD
server initialized successfully
40427: Phone will send a 486 (Busy Here) SIP response if the reject soft key is
used after DND is enabled and disabled
40574: Phone ignores 'Require: 100rel' header in INVITE
40593: 2-way audio (call made from Shared line) gets lost after cancelling transfer
once the far end has performed hold/resume (or cancelled transfer/conf).
40598: Original call does not get resumed when transfer attempt is cancelled by
pressing the active termination key in certain call scenarios.
40669: Caller ID using up.useDirectoryNames="1" stops working when sip and so
logs set at 0
40686: DTMF tones are transmitted in band when RFC 2833 is negotiated on a
SoundStation IP 4000
40694: When call is put on hold at shared line the soft keys "New Call", Transfer",
"Conf", "More" don't appear
40724: SoundStation IP 4000: Call Waiting Tone echo‟d to far end caller.
40804: When new call arrives while user is in the USB Recording „play‟ screen but
not playing audio, incorrect softkeys are displayed
41199: 802.1x packets do not get forwarded by SoundPoint IP 320, 330, 430, 550,
560, 650 phones
41355: Phone responds with 501 to UPDATE request, which it should not do.
41364: Phone does not honor MIME Type for Telephone-Event in SDP Answer
Page 76 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
.cfg File
Action
Parameter
Description
sip
added
voice.codecPref.IP_(6|7)000.*
Codec support for IP 6000 and IP
7000.
sip
added
voice.gain.(r|t)x.analog.*.IP_(6|7)000
Gain levels for IP 6000 and IP 7000.
sip
added
voice.gain.(r|t)x.analog.*.IP_6000
Gain levels for IP 6000.
sip
added
voice.(r|t)xEq.hf.IP_(6|7)000.(pre|post)Filte
r.enable
Prefilter and postfilter enable for IP
6000 and IP 7000.
sip
changed
dir.corp.backGroundSync
Changed from 1 to 0, disabling
background sync.
sip
changed
dir.corp.backGroundSync.period
Changed value from 43200 (12
hours) to 86400 (24 hours).
sip
removed
bg.ranges
sip
changed
bg.color.selection
Defines which background is used.
Default is “1,1”. First (left) index is the
type of background. Second is the
index into the table of that type.
Index
Type
1
Predefined backgrounds
2
Solid patterns
3
User-defined bitmaps
sip
added
bg.hiRes.color.pat.solid.*.(name|red|green|
blue)
Defines the name and colour of solid
backgrounds.
sip
added
bg.hiRes.color.bm.*.(em.)?name
Defines colour backgrounds for the
phone‟s display and the expansion
modules‟ displays (em).
sip
added
button.color.selection.*.*.modify
Defines the transform applied to the
button image used for line keys and
soft keys. The two indexes operate as
defined above in bg.color.selection.
The value comprises a transform
method, and parameters for the
transform. Two transforms are
supported – rbgHiLo and none. The
rgbHiLo has six parameters. The first
two apply to the red channel, the next
two to the green and the last to the
blue. The first parameter of a pair
defines the value to use for the
brightest pixels of the button graphic.
The second parameter of a pair
defines the value to use for the
darkest pixels. Intermediate values
are scaled between the pair.
2.25.4 Configuration File Parameter Changes
Copyright © 2011 Polycom, Inc. Page 77
Release Notes - SIP Application Changes
sip
added
bg.hiRes.gray.(pr|bm).*.adj
Defines the adjustment applied to
backgrounds when displayed on a
gray hiRes phone. “pr” in the
parameter name refers to the
predefined background table. “bm”
refers to the user-defined bitmaps
table. The index is the index into the
respective table.
The value is the number of steps to
brighten the image (negative values
darken the image). Each step is 1/16th
of full scale.
sip
added
bg.hiRes.gray.bm.*.name
Defines gray-scale backgrounds for
the phone‟s display and the
expansion modules‟ displays (em).
sip
added
button.gray.selection.*.*.modify
See button.color.selection.*.*.modify
above.
sip
added
bitmap.IP_7000.*.name
Defines the bitmaps used in the user
interface of the IP 7000 phone. This is
the same format as used with other
SPIP phones.
sip
added
ind.anim.IP_7000.*.frame.*.(bitmap|
duration)
Defines the animations used by the IP
7000 phone. This is the same format
as used with other SPIP phones.
sip
added
ind.gi.IP_7000.*.(index|class|physX|physY|
physW|physH)
Defines the graphical indications used
by the IP 7000 phone. This is the
same format as used with other SPIP
phones.
sip
added
log.level.change.(clink|pnetm|peer)
Three new logging types have been
added. “clink” logs low-level Clink2
activity in the IP 7000. “pnetm” logs
mid-level Clink2 activity. “peer” logs
high-level activity.
sip
added
ramdisk.nBlocks.IP_650
This controls the number of blocks of
memory devoted to the ramdisk in the
IP 650 phone.
2.26 Version 3.0.1RevB
2.26.1 Added or Changed Features
None
2.26.2 Removed Features
None
2.26.3 Corrections
42034: Phone freezes when booting from TFTP server in certain
scenarios.
Page 78 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
.cfg File
Action
Parameter
Description
sip
change
dir.corp.backGroundSync.period
Changed value from 300 (5 minutes)
to 43200 (12 hours)
42121: SoundPoint IP 550 and 650 phones will not provision using the
„large‟ sip.ld software image. Phone reports “Application does not
support self provisioning”.
2.27 Version 3.0.1 (Limited Distribution – build-id 3.0.1.0032)
2.27.1 Added or Changed Features
40475: Set VLAN Filtering to 'Off' by default
41025: Set default Corporate Directory background re-sync period to 12
hours
2.27.2 Removed Features
35285: Add check for user part of check-sync. This was causing issues
with the use of Check-Sync for remote re-boot of phones.
2.27.3 Corrections
36320: Echo is heard on handset to handset call during single talk
setting hsAec to 1 on IP650/550/430/330
38960: Enhance packet loss handling on IP 650 to match performance of
IP 601 in large packet loss situations.
39330: DHCPINFORM should apply if boot server address is Null or
0.0.0.0. (0.0.0.0 checking was not working correctly).
39430: Port component in refer-to target URI is needed in a certain
situation
40121: VLAN tag not added to frame that is an IP fragment with between
1 and 3 octets of payload
2.27.4 Configuration File Parameter Changes
Table 2-1
2.28 Version 3.0.0
** Indicates a feature that requires a license-key to be enabled.
2.28.1 Added or Changed Features
**26088: Add RTCP reporting via SIP protocol according to RFC draft
draft-ietf-sipping-rtcp-summary - ) – all supported phone models except
SoundPoint IP 301
**29851: Support Statistics gathering and reporting for QOS monitoring
according to RFC3611 (RTCP-XR) – all supported phone models except
SoundPoint IP 301
Copyright © 2011 Polycom, Inc. Page 79
Release Notes - SIP Application Changes
**30091: Add a Conference Management User Interface for conferences
hosted locally on the phone (SoundPoint IP 550, 560, 650 phones)
**30099: Add uaCSTA support
30134: Allow speakerphone to be disabled by configuration file
30993: "Submit" from Web Browser should not initiate a reconfig/restart
when no changes have been made on the phone.
31442: Make automatic resume on centralized conference optional.
Implemented for uaCSTA purposes; configured using
call.disableAutoResumeCentralConference
**31576: Add 4-way local conferencing on SoundPoint IP 550, 560, 650
phones
**32054: Integrate with corporate directories using LDAP and Active
Directory
32058: Add configurable behavior to support “Single Keypress
Conference Set-up”. Uses call.singleKeyPressConference parameter.
32223: Add sound effects to accompany USB device insertion and
removal
**32848: Add call recording and playback on USB flash drive. Refer to
Technical Bulletin 38084 for details on supported USB devices.
33230: Add SCA Bridging for BroadWorks. Refer to Technical Bulletin
33230 for more details.
34949: Add support for min-expires header.
35150: Add electronic hook-switch capability using Jabra DHSG
protocol on SoundPoint IP 320, 330, 430, 550, 560, 650 phones. This
feature requires BootROM 4.1.0 to operate. Refer to technical bulletin
35150 for more details.
37159: Handle MIME type application/vq-rtcpxr in SIP stack
37256: Jabra Jx10 electronic hook switch support on SoundPoint IP 320,
330, 430, 550, 560, 650 phones. Requires an “Interface Cable” from the
headset base to the phone for use. Refer to technical bulletin 35150 for
more details.
**37551: Add enhanced speed dial capability.
38443: Support full complement of BLF parties on SoundPoint IP 650
plus 3 EMs using UDP
38847: Line-Key and Soft-Key Labels changed to white text with 3-D
appearance on SoundPoint IP 550, 560, 650 phones.
38979: Make UI background bitmap configurable – SoundPoint IP 550,
560 and 650 phones
39071: DHCPINFORM should apply if boot server address is null
Page 80 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
39072: Reduce DHCPINFORM retry timeouts
39305: Increase Handset transmit loudness by 3dB to better meet
standards AS/NZS 60950 and AS/ACIF S004, as directed by Category
C33 of the Telecommunications Labeling Notice (TLN) (for Australia).
39330: DHCPINFORM should apply if boot server address is 0.0.0.0
39344: Update XML Dictionaries for SIP 3.0.0
39695: Lower minimum syslog.renderLevel to 0 (from 1)
2.28.2 Removed Features
37321: Remove support for Asian languages from IP 600 and IP 601 phones
(due to memory limitations)
2.28.3 Corrections
30170: Icon Frame is missing when pressing menu key
30814: Phone sends INVITE with an incomplete SDP section in a certain call
sequence.
30903: Packet Loss statistics „jump‟ if calls are transferred.
30990: LED does not blink for incoming call on IP 301 when DND enabled and
call.rejectBusyOnDnd=0.
32668: When a call on shared line is put on hold, pressing and holding line key
of a remote shared line causes incorrect soft keys to appear.
34445: Do Not Disturb feature fails on cancellation of second incoming call
when call.rejectBusyOnDnd=0.
35459: On configuring "Identification - Auth Password" in web interface for
configuration, the parameter value is entered in override mac-phone.cfg
35937: SoundPoint IP 550,560,650 phones do not support setting Tx Digital
gain in config file
35963: Large XHTML document can trigger reboot on phones with more than
16MB RAM
36063: HD-Voice Handsets are marginal with respect to hearing aid
compatibility (HAC)
36296: Dialing from directory or hot-dialing bypasses automatic off-hook call
placement
36490: Display Diagnostics has some areas that do not work correctly.
36583: IP 301 logs ssps errors during bootup and when establishing a
handsfree call
36677: IP320/330 does not update its Presence status when a roaming buddy
changes their status
36680: Dial tone can become momentarily very loud when cancelling conf call
Copyright © 2011 Polycom, Inc. Page 81
Release Notes - SIP Application Changes
36751: EM display diagnostics fails during hot plug-in
37071: Internal per-line call limit can be overridden on platforms that do no
allow 24 calls per line
37111: "Using default certs" log message appears when configuring for
"Custom cert" only
37116: Date and time disappear from the phone's idle screen when browsing
menu during call
37184: Digest Authentication Password used for downloading configuration
files is displayed in log files
37227: The registration icon disappears when IP301 establishes a conference
call
37391: Phone does not start correctly if the contact directory XML syntax is
not correct
37420: SIP Server Fall-back --- IP 320 and IP 330 -- Line Information screen
does not show the server info when 3rd SIP server becomes the working
server.
37426: Cannot change selection in Clock Time menu more than once without
exiting
37428: Selecting another language forces exit from language menu
37603: Key remapping does not show correct values in diagnostics menu on
IP 320, IP 330 and IP 4000
37679: File TX Tries setting in flash could be set to invalid value 0
37690: Phone does not retry ACK when receiving duplicate 200 OK
37709: SoundPoint IP 320 and IP 330 phones may re-boot after several days
when the idle micro-browser is configured and active.
37711: Brief audio „noise‟ due to SRTP encryption key change.
37719: Pressing Resume soft key on phone after sending an unresolvable
hostname during a blind transfer reboots or freezes the far end phone
37726: DNS SRV queries can incorrectly append search domain when it is
already present
37851: SRTP phone doesn't include crypto suite in group pickup signaling
37855: Join soft-key is not available when maximum call appearances are
used
37906: IP301 does not show watch buddy icon when peer-to-peer watch buddy
is enabled
37915: Peer-to-Peer Presence: Blocking contact in Watcher List creates extra
contact "SPIP" in directory menu
38021: Ringer type 12 is not playing correctly
Page 82 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
38219: While receiving multiple NOTIFY messages ,the phone may not send an
invite to initiate a call.
38279: If a 403 response is received by the phone when attempting to
complete a call transfer in the proceeding state the phone may re-boot.
38308: Packet Loss count does not increment correctly when a Held call is
resumed and the SSRC value changes.
38334: MKI format in RTP and RTCP packets is incorrect
38540: Packet channel statistics computation not resetting properly when
SSRC changes
38732: Line status icon does not change back on line 2 after being on speaker
or handset – SoundPoint IP 330/320
38902: UI malfunctions when remote shared line is in hold status and local
phone attempts a new call
39041: Icon may indicate phone is unregistered after successful re-registration
if voIpProt.SIP.serverFeatureControl.cf=1 or
voIpProt.SIP.serverFeatureControl.dnd=1
39074: Microbrowser: clicking a link to non-responsive server takes a long
time to timeout
39184: Read-only directory can be edited on IP 320 and IP 330 if phone is in
digit collection state when contact directory is opened
39338: Some of the SRTP session parameters are incorrectly spelled in the
SDP (e.g. UNENCRYPTED_SRTCP is represented as UNENCRYPTED_RTCP)
39362: Phone does not play incoming RTP when offered send-only stream.
39419: Maximum Backlight Intensity setting has very little effect on
SoundPoint IP 560 phones.
39431: Display Diagnostics shows very minimal changes on the display on IP
550 and IP 650
39438: Backlight does not update immediately after pressing cancel on the
maximum intensity screen
39490: In some call scenarios the phone may not display the SRTP secure line
icon even though the call is encrypted.
39502: DigitMap: The + character does not get matched in a dial plan.
39601: In IP 320 and IP 330 phone's local contact edit menu, cursor flashes on
the character just entered instead of after the character
39618: font500Prop_16_U0000_U00FF.fnt has anomalously wide "K"
39629: When reg.1.callsPerLineKey=1 is set, and a conference is established
while transferring the call, the phone hangs and reboots
39631: Idle browser cuts volume icon
39652: Some layered windows are incorrectly clipped
Copyright © 2011 Polycom, Inc. Page 83
Release Notes - SIP Application Changes
.cfg File
Action
Parameter
Description
sip
added
voIpProt.SDP.
useLegacyPayloadTypeNegotiation
sip
added
voIpProt.SIP.csta
Enables uaCSTA.
sip
added
up.handsfreeMode
Enables or disables hands-free
speakerphone.
phone1
added
up.analogHeadsetOption
Selects optional external hardware for
use with a headset attached to the
phone's analog headset jack.
sip
changed
tone.chord.callProg.6.offDur
Changed from 0 to 10000.
sip
changed
tone.chord.callProg.6.repeat
Changed from 1 to 2.
sip
changed
se.pat.ringer.12.name="Ringback-style"
Added 100ms of silence to start of
pattern.
sip
removed
voice.gain.rx.analog.handset.wideband
voice.gain.rx.analog.handset.sidetone.
wideband
voice.gain.tx.analog.handset.wideband
voice.handset.wideband
voice.handset.wideband.rxdg.adjust
Controlled gain for wideband handset.
This control is now performed through
the parameters that do not include
“.wideband”.
sip
added
voice.qualityMonitoring
The voice.qualityMonitoring section
controls the Voice Quality Monitoring
feature.
sip
added
tcpIpApp.keepalive.tcp.idleTransmitInterval
tcpIpApp.keepalive.tcp.
noResponseTrasmitInterval
tcpIpApp.keepalive.tcp.sip.tls.enable
Controls TCP keep-alive on SIP TLS
connections.
sip
added
call.singleKeyPressConference
call.localConferenceCallHold
Enables new conference behaviors.
sip
added
call.disableAutoResumeCentralConference
For use with uaCSTA feature for
centralized confrerencing.
sip
added
bg.hiRes.gray.pat.solid.x.name
bg.hiRes.gray.pat.solid.x.red
bg.hiRes.gray.pat.solid.x.green
bg.hiRes.gray.pat.solid.x.blue
bg.hiRes.gray.bm.x.name
Sets up color (gray-scale) and
graphical backgrounds for IP 550,
IP560 and IP 650 phones.
sip
added
feature.x.name
Added new features “nwayconference”, “call-recording” and
“corporate-directory”
phone1
added
reg.x.bargeInEnabled
Enables barge in feature for SCAs.
sip
added
dir.corp
The dir.corp section controls the
Corporate Directory feature.
sip
added
usb.set1.device.1.vendor
usb.set1.device.1.product
Identifies supported USB devices.
This list should be populated only with
devices that are known to work with
the phones. See Technical Bulletin
38084 for details.
2.28.4 Configuration File Parameter Changes
Table 2-2
2.29 Version 2.2.2
2.29.1 Added or Changed Features
35534: De-couple Presence Signaling from Idle Screen Soft-key UI
Page 84 Copyright © 2011 Polycom, Inc.
36931: Add support for SoundPoint IP 560 product.
Release Notes - SIP Application Changes
37053: Add ability to make local contact directory read-only from the
phone
38328: Add check for local contact directory changes during
configuration change checks
38357: Add ability to adjust the maximum brightness of the SoundPoint
IP 550 and 650 phones.
38371: Allow for TCP keep-alive on SIP signaling TLS connections
38654: Add support for SoundPoint IP 320 Part Number 2345-12200-005
and SoundPoint IP 330 Part Number 2345-12200-004 for China market.
38888: Add ability to adjust the maximum brightness of SoundPoint IP
Backlit Expansion Modules.
2.29.2 Removed Features
38813: Remove 1000 half duplex as a valid ethernet configuration.
2.29.3 Corrections
34800: MWI Notify: Message Waiting Counts are ignored if "Messages-
Waiting" is set to "no"
35692: Functionality breaks down on pressing "conference>>cancel"
soft keys after transfer try is rejected. Phone reboots.
36566: Microbrowser: Left arrow when on first field in a form makes
cursor turn invisible
36786: Changing audio modes (e.g. handsfree to handset) during call
set-up mode may not work correctly in some circumstances.
37284/37661: During a Blind Transfer the phone should terminate the
call on receipt of a 180 Ringing Response.
37313: RTP packet size incorrect when SRTP authentication turned off
37316: Authentication failing when phones have different payload size
37334: Disabling CDP from the phone menu causes an unnecessary
reboot
37709: SoundPoint IP 330/320 phones using the idle micro-browser may
re-boot after several days due to low memory.
38112: Logging message indicates that default cert bundle in use when
custom only has been selected.
38344: If URL-dialing is disabled in the configuration file, the phone
shows Number@ServerIP for caller ID (This issue occurs on SIP 2.2.0
and SIP 2.2.1 releases only).
38430: In a BLA configuration attempting to make a call on a remotely
busy shared line may cause the phone to re-boot instead of displaying
Copyright © 2011 Polycom, Inc. Page 85
Release Notes - SIP Application Changes
.cfg File
Action
Parameter
Description
sip
added
tcpIpApp.keepalive.tcp.
idleTransmitInterval
Sets the interval of the TCP keepalive packets.
sip
added
tcpIpApp.keepalive.tcp.
noResponseTrasmitInterval
Set the retransmission interval when
the server fails to acknowledge the
TCP keep-alive.
sip
added
tcpIpApp.keepalive.tcp.sip.tls.
enable
Enables sending a TCP keep-alive
packet from the phone to the server.
The server is expected to respond
with a TCP keep-alive ack. This is
only used with TLS sessions.
sip
added
dir.local.readonly
When set to “1”, the contact directory
cannot be changed and
[MACADDRESS]-directory.xml is not
uploaded.
sip
added
pres.idleSoftKeys
If set to “0”, appearance of presence
idle soft keys is disabled.
“Service Unavailable”. Occurs on SoundPoint IP 330/320, 430, 550, 650
phones.
38435: When the phone's local directory is writable, unable to add a new
contact by selecting "new entry" on SoundPoint IP 330/320 phones.
38666: If a call is initiated in hands-free mode and the Ringback Tone is
server generated the far-end user may experience echo when they
answer the call. If the originating phone is switched to handset mode
and back to hands-free mode the echo goes away. Occurs on
SoundPoint IP 330/320, 430, 550, 650 phones.
38678: In a particular network configuration when using BLA the
bridged line indication does not light up properly due to a missing
NOTIFY from the phone.
2.29.4 Configuration File Parameter Changes
2.30 Version 2.2.1 (Limited Release)
2.30.1 Added or Changed Features
38371: When SIP over TLS is configured the phone will send TCP Keep-
Alive messages to the SIP server every 30 seconds, and will retry 3
times (at 20 seconds) before resetting (RST) the connection if no
response is received
2.30.2 Removed Features
None.
2.30.3 Corrections
36557: When SRTP is enabled and “so” logging level is set to 1, the
RTCP sender report displays encrypted values in the log file
Page 86 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
.cfg File
Action
Parameter
Description
sip
added
sec.srtp.offer.HMAC_SHA1_80
If set to 1 or Null, a crypto line with
the AES_CM_128_HMAC_SHA1_80
crypto-suite will be included in offered
SDP.
If set to 0, the crypto line is not
included.
sip
added
sec.srtp.offer.HMAC_SHA1_32
If set to 1, a crypto line with the
AES_CM_128_HMAC_SHA1_32
crypto-suite will be included in offered
SDP.
If set to 0 or Null, the crypto line is not
included.
37651: RTP Timestamp not updated correctly for silence packets
37690: Phone does not retry ACK when receiving duplicate 200 OK
37708: Phones fail SIP TLS registration when SNTP server is not
configured
37851: SRTP phone doesn't include Crypto Suite in Group Pickup
signaling
37873: Crypto line in answer does not have correct tag field
37878: Multiple crypto suites not handled when there is a re-INVITE
37879: SRTCP packets have invalid authentication tags
37968: Phone with multiple lines using TLS not re-registering on loss of
connection
38110: Far end hears noise when an SRTP call is taken off hold with
some SIP servers
38249: SRTP lifetime value cannot be parsed correctly by the called
party
38384: During a local SRTP conference, a far end holding then resuming
may result in one-way audio or noise with some SIP servers
2.30.4 Configuration File Parameter Changes
2.31 Version 2.2.0
2.31.1 Added or Changed Features
22532: When there has been no activity in a menu for a configurable
period of time, the phone returns to the idle display. This does not
happen if the user is entering data using a menu.
25274: Added sending vendor identifier information through DHCP
25702: Added microbrowser support for accepting and displaying a URL
that points directly to a BMP image (previously it was necessary to
embed BMP images in an XHTML document)
Copyright © 2011 Polycom, Inc. Page 87
Release Notes - SIP Application Changes
27040: Added new configurable ring-while-busy options
28029: Added microbrowser support for two-dimensional table
navigation using all four arrow keys
28747: Added a general flash file system caching mechanism so that
downloaded resources can be stored in non-volatile memory
29030: Added automatic provisioning support for individual image files
29854: Added support for tracking of missed calls to be configurable on
a per-line basis
31558: Added synchronization of local DND/CF features with server-
based DND/CF features
31840: Set transfer time-out for image file download to worst case
scenario
32259: Added microbrowser support for recognizing mime types
32648: Reformatted call list entries
33616: Added configuration option for default transfer type for
SoundPoint IP 320 and 330 phones
33748: Improved resistance to denial of service attacks aimed at
phone‟s web server
34131: Changed URL dialing terminology from "Name" to "URL"
34434: Implemented 300Hz high pass transmit filter to reduce low
frequency noise (noise creates problems in some network line echo
cancellers). This can be enabled or disabled.
34573: Added support for re-establishing a TLS connection if the
connection closes
34625: Added ability to discover provisioning server address using
DHCPINFORM
34651: Added phone serial number (MAC address) to user-agent string
HTTP Gets
34685: Renamed "Services" menu entry to "Applications"
34705: Added support in microbrowser for form functionality when
embedded in tbody or out of tbody
34707: Added low-delay handset acoustic echo canceller for SoundPoint
IP 320, 330, 430, 550 and 650 phones. This can be enabled or disabled.
34874: If all DNS servers are found to be unreachable, the phone
suppresses DNS queries for 5 minutes (as per RFC 2308 Sec 7.1)
34998: Increased maximum number of registrations on SoundPoint IP
650 phones to 34
35039: Pressing "Exit" soft key when using the microbrowser should
return user to telephony application
Page 88 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
35040: Added configurable timeout parameter to allow microbrowser to
return to telephony application after a period of inactivity in the
microbrowser
35043: Added configurable option to display or hide browser status
messages in microbrowser
35087: Changed boot-up behaviour so that idle browser only starts
about 2 minutes after the phone has booted up (this is to optimize
memory use)
35099: Added support for TLS transport to Syslog
35199: Improved some translations in Norwegian XML dictionary file
35285: Add check for user part of check-sync
35296: Added support for managing TLS custom certificates via the
configuration file system
35311: Added support for specifying different versions of the
application executable and configuration files in the <Ethernet
address>.cfg file on the boot server
35372: Pressing the “Exit” function key on the SoundStation IP 4000
phone when using the microbrowser should return user to telephony
application
35373: Changed appearance of soft keys when running microbrowser so
that they look the same as when running the telephony application
35419: Added user interface for configuring no-answer and busy
forwarding behavior
35481: Added support for Backlit Expansion Module
35507: Adding configuration parameter to control the timeout back to
the idle display after a period of inactivity in a menu
36030: Implemented Ethernet ingress filtering for DoS suppression and
VLAN filtering
36277: Added ability to delete the contact number entered in the
Forward menu
36531: Updated all translation dictionary files to rename "Services"
menu entry to "Applications"
2.31.2 Removed Features
36079: Removed support for the SoundPoint IP 300 and 500 phones
2.31.3 Corrections
24021: Call display gets corrupted in IP-dialed call if caller presses a
digit then puts call on hold
25744: Spaces go missing in text in microbrowser occasionally
Copyright © 2011 Polycom, Inc. Page 89
Release Notes - SIP Application Changes
26110: Volume level cannot be changed in audio diagnostics mode
26231: ACD login failure should cause busy tone to be played
26389: Forward contact which has been disabled is not displayed after a
reboot
26935: ACD icon not suppressed if feature is disabled in sip.cfg but
activated in phone1.cfg
27105: The idle browser occasionally displays when the menu is being
updated
27958: Phone hears busy tone for 2 seconds after far end hangs up and
another call is already in the incoming state and has triggered the call
waiting alert
28419: Divert settings for lines 7 to 12 are not used
28503: When in the “held” state, a shared line hears ring tone instead of
call waiting tone when another call comes in
28570: Stuttered dial tone (indicating voice mail waiting) does not work
on shared line
28622: Some UNICODE ranges are not properly mapped
28681: "Forward" is not removed from menu when function disabled
29014: Cannot edit the local directory on the phone if the file is corrupt
on the server
29358: Phone may malfunction/reboot if the specified DNS server is
down and an invalid SNTP address is configured
29470: Cursor is in wrong position when performing a factory reset on
the SoundPoint IP 301 phone
29573: Phone may freeze if a DNS server address is all zeroes
29966: Phone may reboot if incorrect information is entered in the menu
for custom CA certificate
30880: Phone may malfunction/reboot when editing a server address
which is 255 characters long
30902: Auto reject or divert settings changed in a contact after entering
contact directory by pressing and holding a speed dial line key are not
correctly displayed when next pressing and holding that speed dial line
key
31019: There is no confirmation pop-up message after choosing to reset
the local security key
31326: Transferring a call to windows messenger or office
communicator may leave the phone in a frozen state
31886: Remote resume does not work on BLA line when call between
two other phones sharing the same line has been put on hold
Page 90 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
31994: Trying to delete a null unicode character in the contact list
causes the phone to lock-up/reboot.
32179: When SAS-VP provisioning is used, the boot server password is
visible in the application log file
32816: Phone may lock-up on subsequent call if using NTLM and
received transfer from a non-NTLM phone
32476: IP601 does not work correctly when Presence feature is enabled
with LCS server without using Roaming Buddies
33105: "Hold" does not work if selected just before a Conference is
completed
33748: Web server has vulnerability to DOS attacks
33931: Not all keys on phone can be remapped to Null
34089: SoundPoint IP 430 phone keeps rebooting if a function key is
remapped to null in the configuration files
34196: Phone keeps rebooting when SIP server address is not a fully
qualified domain name and primary DNS server replies to queries with
ICMP destination unreachable packets (due to service being turned off)
and secondary DNS server is not configured with NAPTR and SRV
entries for the SIP server
34237: Default directory file (000000000000-directory.xml) is not
downloaded by the phone when the <Ethernet-address>-directory.xml
file does not exist on the boot server
34258: Log file is deleted when it reaches the configured size limit even
though log.render.file.upload.append.limitMode is set to “stop”
34271: SoundPoint IP 430/550/650 phones may reboot when
microbrowser XHTML page contains combined FORM and TABLE
elements
34460: Local directory file larger than 10kB is downloaded by phone
once but on subsequent reboots the phone freezes
34578: Phones may lock-up when downloading a directory file which
contains an empty contact field
34636: Call on a shared line may lose audio when cancelling a transfer
after the far end has already cancelled a transfer or conference
34641: Emergency Call Routing does not work correctly if multiple
numbers are configured in a single entry in the configuration file e.g.
dialplan.1.routing.emergency.1.value=911,9911
34649: First call after a reboot may demonstrate one-way audio if
phones have different codec preferences and
voIpProt.SDP.answer.useLocalPreferences parameter is set to default
34891: SoundStation IP 4000 loudness does not decrease for bottom six
volume settings
Copyright © 2011 Polycom, Inc. Page 91
Release Notes - SIP Application Changes
.cfg File
Action
Parameter
Description
sip
added
voIpProt.SIP.csta
Not currently used, will be used in a
future release.
sip
added
voIpProt.SIP.serverFeatureControl.d
nd
See Administrator‟s Guide for SIP
2.2.0 for details
sip
added
voIpProt.SIP.serverFeatureControl.c
f
See Administrator‟s Guide for SIP
2.2.0 for details
sip
added
up.toneControl.bass
Not currently used, will be used in a
future release.
sip
added
up.toneControl.treble
Not currently used, will be used in a
future release.
35320: If two function keys are remapped to dial specific speed dial
numbers, only the first one will work
35480: SoundPoint IP 320 and 330 phones allow watching only 7
buddies instead of 8 and may lock-up when an 8th watched buddy is
added
35490: SoundPoint IP 320 and 330 phones do not display SAS-VP failure
messages during boot-up
35879: Nonce counter not incremented in PRACK
36031: If a phone is configured to use TLS for the 2nd line and TCP for
the 1st, the 2nd line does not register
36107: SoundStation IP 4000 phone drops maximum size packets when
VLAN is enabled
36477: Configuring the nat.signalPort parameter may cause the phone
to lock-up
36775: Route-Set susceptible to change mid-dialog in certain situations
36882: Selecting a speed dial number using the „nn#‟ key sequence
does not work on SoundPoint IP 320 and 330 phones when the phone is
unregistered or is using URL dialing mode
36905: CDP packet always advertises LAN duplex mode as "Duplex:
Full"
36948: On SoundPoint IP 320 and 330 phones, if the Dial and Menu keys
are pressed at the same time after entering digits from the idle display,
incorrect soft keys are displayed
36967: If the phone receives an INVITE with SDP which contains video
information, it returns a malformed response
37086: Phone ignores expiration date of CA certificate if SNTP is only
set via DHCP
37632: Out of order SCA signaling can lead to improper handling of
Shared Lines in some situations.
37646: DNS SRV querying after A record cache makes registration fail
2.31.4 Configuration File Parameter Changes
Page 92 Copyright © 2011 Polycom, Inc.
Release Notes - SIP Application Changes
.cfg File
Action
Parameter
Description
sip
added
up.audioSetup.auxInput
Not currently used, will be used in a
future release.
sip
added
up.audioSetup.auxOutput
Not currently used, will be used in a
future release.
sip
added
up.idleTimeout
See Administrator‟s Guide for SIP
2.2.0 for details
sip
added
se.pat.ringer.12.inst.5.type="branch"
se.pat.ringer.12.inst.5.value="-4"
sip
added
voice.txPacketFilter
See Administrator‟s Guide for SIP
2.2.0 for details
sip
added
voice.codecPref.IP_7000.xxx
Not currently used, will be used in a
future release.
sip
added
voice.audioProfile.Lin16.frequency
voice.audioProfile.G7221.xxx
voice.audioProfile.G7221C.xxx
voice.audioProfile.Siren14.xxx
voice.audioProfile.Siren22.xxx
Not currently used, will be used in a
future release.
sip
added
Several gain and other voice
parameters have been added.
The entire gain section in sip.cfg must
be updated. Failure to do this will
affect the audio performance of the
phone.
sip
added
voice.rxEq.hf.IP_7000.xxx
voice.txEq.hf.IP_7000
Not currently used, will be used in a
future release.
sip
added
call.dialtoneTimeOut
See Administrator‟s Guide for SIP
2.2.0 for details
sip
added
call.disableAutoResumeCentralConf
erence
Not currently used, will be used in a
future release.
sip
added
call.singleKeyPressConference
Not currently used, will be used in a
future release.
sip
added
call.transfer.blindPreferred
See Administrator‟s Guide for SIP
2.2.0 for details
Sip
added
call.cellPhoneAutoBridging
Not currently used, will be used in a
future release.
Sip
added
bitmap.IP_7000.xxx
Not currently used, will be used in a
future release.
Sip
added
log.level.change.srtp
See Administrator‟s Guide for SIP
2.2.0 for details
Sip
added
log.level.change.clink
log.level.change.pnetm
log.level.change.peer
Not currently used, will be used in a
future release.
Copyright © 2011 Polycom, Inc. Page 93
Release Notes - SIP Application Changes
.cfg File
Action
Parameter
Description
Sip
added
sec.srtp.enable
sec.srtp.leg.enable
sec.srtp.offer
sec.srtp.require
sec.srtp.key.lifetime
sec.srtp.mki.enabled
sec.srtp.sessionParams.noAuth.offe
r
sec.srtp.sessionParams.noAuth.req
uire
sec.srtp.sessionParams.noEncrypR
TP.offer
sec.srtp.sessionParams.noEncrypR
TP.require
sec.srtp.sessionParams.noEncrypR
TCP.offer
sec.srtp.sessionParams.noEncrypR
TCP.require
sec.srtp.sessionParams.leg.noAuth.
offer
sec.srtp.sessionParams.leg.noAuth.r
equire
sec.srtp.sessionParams.leg.noEncry
pRTP.offer
sec.srtp.sessionParams.leg.noEncry
pRTP.require
sec.srtp.sessionParams.leg.noEncry
pRTCP.offer
sec.srtp.sessionParams.leg.noEncry
pRTCP.require
sec.srtp.sessionParams.IP_4000.no
Auth.offer
sec.srtp.sessionParams.IP_4000.no
Auth.require
sec.srtp.sessionParams.IP_4000.no
EncrypRTP.offer
sec.srtp.sessionParams.IP_4000.no
EncrypRTP.require
sec.srtp.sessionParams.IP_4000.no
EncrypRTCP.offer
sec.srtp.sessionParams.IP_4000.no
EncrypRTCP.require
sec.srtp.leg.allowLocalConf
See Technical Bulletin 25751 for
details.
sip
added
license.polling.time
See Administrator‟s Guide for SIP
2.2.0 for details
sip
added
feature.16.name
feature.16.enabled
Not currently used, will be used in a
future release.
sip
added
mb.main.idleTimeout
See Administrator‟s Guide for SIP
2.2.0 for details
sip
added
mb.main.statusbar
See Administrator‟s Guide for SIP
2.2.0 for details
sip
added
pnet.role
Not currently used, will be used in a
future release.
Page 94 Copyright © 2011 Polycom, Inc.
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