Polycom Personal Computer 3804 11530 222 User Manual

Release Notes

SIP Application

SoundPoint and SoundStation IP

Version 2.2.2

2 December 2007

Part Number 3804-11530-222

Release Notes - SIP Application

Table of Contents

1. GENERAL................................................................................................................................... 1

1.1

1.2

1.3

IMPORTANT NOTES ................................................................................................................ 1

SYSTEM REQUIREMENTS........................................................................................................ 1

DISTRIBUTION FILES .............................................................................................................. 2

2. CHANGES................................................................................................................................... 3

2.1

2.1.1

VERSION 2.2.2 ....................................................................................................................... 3

Added or Changed Features......................................................................................... 3

Removed Features......................................................................................................... 3

Corrections ................................................................................................................... 3

Configuration File Parameter Changes ....................................................................... 4

VERSION 2.2.1 (LIMITED RELEASE) ....................................................................................... 4

Added or Changed Features......................................................................................... 4

Removed Features......................................................................................................... 4

Corrections ................................................................................................................... 5

Configuration File Parameter Changes ....................................................................... 5

VERSION 2.2.0 ....................................................................................................................... 5

Added or Changed Features......................................................................................... 5

Removed Features......................................................................................................... 7

Corrections ................................................................................................................... 7

Configuration File Parameter Changes ..................................................................... 10

VERSION 2.1.2 ..................................................................................................................... 14

Added or Changed Features....................................................................................... 14

Removed Features....................................................................................................... 14

Corrections ................................................................................................................. 14

Configuration File Parameter Changes ..................................................................... 15

VERSION 2.1.1 C.................................................................................................................. 16

Added or Changed Features....................................................................................... 16

Removed Features....................................................................................................... 16

Corrections ................................................................................................................. 16

Configuration File Parameter Changes ..................................................................... 17

VERSION 2.1.1 ..................................................................................................................... 17

Added or Changed Features....................................................................................... 17

Removed Features....................................................................................................... 17

Corrections ................................................................................................................. 17

Configuration File Parameter Changes ..................................................................... 19

VERSION 2.1.0 ..................................................................................................................... 20

Added or Changed Features....................................................................................... 20

Removed Features....................................................................................................... 21

Corrections ................................................................................................................. 21

Configuration File Parameter Changes ..................................................................... 23

VERSION 2.0.3 B.................................................................................................................. 24

Added or Changed Features....................................................................................... 24

Removed Features....................................................................................................... 25

Corrections ................................................................................................................. 25

2.1.2

2.1.3

2.1.4

2.2

2.2.1

2.2.2

2.2.3

2.2.4

2.3

2.3.1

2.3.2

2.3.3

2.3.4

2.4

2.4.1

2.4.2

2.4.3

2.4.4

2.5

2.5.1

2.5.2

2.5.3

2.5.4

2.6

2.6.1

2.6.2

2.6.3

2.6.4

2.7

2.7.1

2.7.2

2.7.3

2.7.4

2.8

2.8.1

2.8.2

2.8.3

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Release Notes - SIP Application

2.8.4

Table of Contents

Configuration File Parameter Changes ..................................................................... 25

VERSION 2.0.3 ..................................................................................................................... 25

Added or Changed Features....................................................................................... 25

Removed Features....................................................................................................... 25

Corrections ................................................................................................................. 25

Configuration File Parameter Changes ..................................................................... 26

2.9

2.9.1

2.9.2

2.9.3

2.9.4

2.10 VERSION 2.0.2 ..................................................................................................................... 27

2.10.1

2.10.2

2.10.3

2.10.4

Added or Changed Features....................................................................................... 27

Removed Features....................................................................................................... 28

Corrections ................................................................................................................. 28

Configuration File Parameter Changes ..................................................................... 28

2.11 VERSION 2.0.1 B.................................................................................................................. 28

2.11.1

2.11.2

2.11.3

2.11.4

Added or Changed Features....................................................................................... 28

Removed Features....................................................................................................... 28

Corrections ................................................................................................................. 28

Configuration File Parameter Changes ..................................................................... 28

2.12 VERSION 2.0.1 ..................................................................................................................... 29

2.12.1

2.12.2

2.12.3

2.12.4

Added or Changed Features....................................................................................... 29

Removed Features....................................................................................................... 29

Corrections ................................................................................................................. 29

Configuration File Parameter Changes ..................................................................... 31

2.13 VERSION 2.0.0 (BETA RELEASE ONLY)................................................................................ 32

2.13.1

2.13.2

2.13.3

2.13.4

Added or Changed Features....................................................................................... 32

Removed Features....................................................................................................... 34

Corrections ................................................................................................................. 34

Configuration File Parameter Changes ..................................................................... 37

2.14 VERSION 1.6.7 ..................................................................................................................... 40

2.14.1

2.14.2

2.14.3

2.14.4

Added or Changed Features....................................................................................... 40

Removed Features....................................................................................................... 40

Corrections ................................................................................................................. 40

Configuration File Parameter Changes ..................................................................... 41

2.15 VERSION 1.6.6 C (LIMITED DISTRIBUTION) ......................................................................... 42

2.15.1

2.15.2

2.15.3

2.15.4

Added or Changed Features....................................................................................... 42

Removed Features....................................................................................................... 42

Corrections ................................................................................................................. 42

Configuration File Parameter Changes ..................................................................... 42

2.16 VERSION 1.6.6 B.................................................................................................................. 42

2.16.1

2.16.2

2.16.3

2.16.4

Added or Changed Features....................................................................................... 42

Removed Features....................................................................................................... 42

Corrections ................................................................................................................. 42

Configuration File Parameter Changes ..................................................................... 43

2.17 VERSION 1.6.6 ..................................................................................................................... 43

2.17.1

2.17.2

2.17.3

2.17.4

Added or Changed Features....................................................................................... 43

Removed Features....................................................................................................... 44

Corrections ................................................................................................................. 44

Configuration File Parameter Changes ..................................................................... 45

2.18 VERSION 1.6.5 ..................................................................................................................... 45

2.18.1

2.18.2

Added or Changed Features....................................................................................... 45

Removed Features....................................................................................................... 46

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Release Notes - SIP Application

Table of Contents

2.18.3

2.18.4

Corrections ................................................................................................................. 46

Configuration File Parameter Changes ..................................................................... 47

2.19 VERSION 1.6.4 ..................................................................................................................... 47

2.19.1

2.19.2

2.19.3

2.19.4

Added or Changed Features....................................................................................... 47

Removed Features....................................................................................................... 47

Corrections ................................................................................................................. 47

Configuration File Parameter Changes ..................................................................... 48

2.20 VERSION 1.6.3 ..................................................................................................................... 48

2.20.1

2.20.2

2.20.3

2.20.4

Added or Changed Features....................................................................................... 48

Removed Features....................................................................................................... 49

Corrections ................................................................................................................. 49

Configuration File Parameter Changes ..................................................................... 50

2.21 VERSION 1.6.2 ..................................................................................................................... 50

2.21.1

2.21.2

2.21.3

2.21.4

Added or Changed Features....................................................................................... 50

Removed Features....................................................................................................... 50

Corrections ................................................................................................................. 50

Configuration File Parameter Changes ..................................................................... 50

2.22 VERSION 1.6.1 ..................................................................................................................... 50

2.22.1

2.22.2

2.22.3

2.22.4

Added or Changed Features....................................................................................... 50

Removed Features....................................................................................................... 51

Corrections ................................................................................................................. 51

Configuration File Parameter Changes ..................................................................... 51

2.23 VERSION 1.6.0 (BETA ONLY)................................................................................................ 51

2.23.1

2.23.2

2.23.3

2.23.4

Added or Changed Features....................................................................................... 51

Removed Features....................................................................................................... 52

Corrections ................................................................................................................. 52

Configuration File Parameter Changes ..................................................................... 53

3. NOTES....................................................................................................................................... 54

3.1

3.1.1

UPGRADING ......................................................................................................................... 54

From Version 2.2.1 to 2.2.2........................................................................................ 54

From Version 2.2.0 to 2.2.1........................................................................................ 54

From Version 2.1.2 to 2.2.0........................................................................................ 54

From Version 2.1.1 C to 2.1.2 .................................................................................... 55

From Version 2.1.1 to 2.1.1 C .................................................................................... 55

From Version 2.1.0 to 2.1.1........................................................................................ 55

From Version 2.0.3 to 2.1.0....................................................................................... 56

From Version 2.0.3 to 2.0.3 B..................................................................................... 56

From Version 2.0.2 to 2.0.3........................................................................................ 56

3.1.2

3.1.3

3.1.4

3.1.5

3.1.6

3.1.7

3.1.8

3.1.9

3.1.10

3.1.11

3.1.12

3.1.13

3.1.14

3.1.15

3.1.16

3.1.17

3.1.18

3.1.19

From Version 2.0.1 to 2.0.2........................................................................................ 57

From Version 2.0.0 to 2.0.1........................................................................................ 57

From Version 1.6.7 to 2.0.0........................................................................................ 57

From Version 1.6.6 to 1.6.7........................................................................................ 58

From Version 1.6.5 to 1.6.6........................................................................................ 58

From Version 1.6.4 to 1.6.5........................................................................................ 58

From Version 1.6.3 to 1.6.4........................................................................................ 59

From Version 1.6.2 to 1.6.3........................................................................................ 59

From Version 1.6.1 to 1.6.2........................................................................................ 59

From Version 1.6.0 to 1.6.1........................................................................................ 59

Page iii

Release Notes - SIP Application

3.2 OUTSTANDING ISSUES.......................................................................................................... 60

4. REFERENCE DOCUMENTS................................................................................................. 62

Table of Contents

Page iv

Release Notes - SIP Application

General

1. General

These release notes apply to version 2.2.2 of the SoundPoint IP SIP application.

This release is a patch release that replaces the 2.2.0 release as the latest generally

available (GA) release.

For more information, refer to the documents listed in Section 4.

1.1 Important Notes

• This software release does not include images for the SoundPoint IP 300 and 500

phone models. If deployments utilize a mix of IP 300 and IP 500 phones along with

newer models the steps detailed in technical bulletin 35311 must be followed. The

technical bulletin is available from www.polycom.com/support/voip (Search the

Knowledge Base for 35311).

• Support for encrypted media using SRTP is available in this release. Due to the

significant inter-operability needs when deploying SRTP, this feature is available

when the phones are used with particular call servers and network infra-structure

only. Please contact your solutions provider to establish whether they offer this

feature. Anyone wishing to use this feature for inter-operability testing should contact

Polycom to receive technical bulletin 25751 for details on how to enable this feature.

• This is the first GA release to support the SoundPoint IP 560 product platform.

1.2 System Requirements

Platform

BootROM version

2.6.1 or greater

SoundPoint IP 301

SoundPoint IP 320

SoundPoint IP 330

SoundPoint IP 430

SoundPoint IP 501

SoundPoint IP 550

SoundPoint IP 560

SoundPoint IP 600

SoundPoint IP 601

SoundPoint IP 650

SoundStation IP 4000

3.2.3RevB or greater

3.1.3 or greater

2.6.1 or greater

3.2.3 or greater

4.0.1 or greater

2.6.1 or greater

3.1.0 or greater

3.2.2RevB or greater

3.1.2 or greater

Page 1

Release Notes - SIP Application

General

1.3 Distribution Files

The following files constitute the 2.2.2 distribution of the SoundPoint / SoundStation IP SIP

application. For centrally provisioned systems, copy these files to the boot server,

maintaining the folder hierarchy present in the zip file.

Some of the configuration files must be modified. Refer to the Administrator Guide for

details.

Files

sip.ld

Description

Concatenated SIP application executable, Version

2.2.2.0084 for all platforms

2345-11300-010.sip.ld

2345-12200-002.sip.ld

2345-12200-005.sip.ld

2345-12200-001.sip.ld

2345-12200-004.sip.ld

2345-11402-001.sip.ld

2345-11500-030.sip.ld

2345-11500-040.sip.ld

2345-12500-001.sip.ld

2345-12560-001.sip.ld

2345-11600-001.sip.ld

2345-11605-001.sip.ld

2345-12600-001.sip.ld

2201-06642-001.sip.ld

SIP application executable for SoundPoint IP 301 – Version 2.2.2.0084

SIP application executables for SoundPoint IP 320 – Version 2.2.2.0084

SIP application executables for SoundPoint IP 330 – Version 2.2.2.0084

SIP application executable for SoundPoint IP 430 – Version 2.2.2.0084

SIP application executables for SoundPoint IP 501 – Version 2.2.2.0084

SIP application executable for SoundPoint IP 550 – Version 2.2.2.0084

SIP application executable for SoundPoint IP 560 – Version 2.2.2.0084

SIP application executable for SoundPoint IP 600 – Version 2.2.2.0084

SIP application executable for SoundPoint IP 601 – Version 2.2.2.0084

SIP application executable for SoundPoint IP 650 – Version 2.2.2.0084

SIP application executable for SoundStationt IP 4000 – Version

2.2.2.0084

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

000000000000-directory~.xml

example master configuration file

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 (no IP 30X support):

Chinese, China (for IP 6XX, IP 550, 560 and IP 4000 only)

Danish, Denmark

Dutch, Netherlands

English, Canada

English, United Kingdom

English, United States

French, France

German, Germany

Italian, Italy

Japanese, Japan (for IP 6XX, IP 550 and IP 4000 only)

Korean, Korea (for IP 6XX, IP 550 and IP 4000 only)

Norwegian, Norway

Portuguese, Portugal

Russian, Russia

Spanish, Spain

Swedish, Sweden

SoundPointIPWelcome.wav

start up welcome sound effect

Page 2

Release Notes - SIP Application

Changes

2. Changes

2.1 Version 2.2.2

2.1.1 Added or Changed Features

• 35534: De-couple Presence Signaling from Idle Screen Soft-key UI

• 36931: Add support for SoundPoint IP 560 product.

• 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.

2.1.2 Removed Features

• 38813: Remove 1000 half duplex as a valid ethernet configuration.

2.1.3 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: 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.

Page 3

Release Notes - SIP Application

Changes

• 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 “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.

2.1.4 Configuration File Parameter Changes

.cfg File

Action

Parameter

Description

Sets the interval of the TCP keep-

alive packets.

sip

added

tcpIpApp.keepalive.tcp.

idleTransmitInterval

tcpIpApp.keepalive.tcp.

noResponseTrasmitInterval

sip

added

Set the retransmission interval when

the server fails to acknowledge the

TCP keep-alive.

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.

When set to “1”, the contact directory

cannot be changed and

tcpIpApp.keepalive.tcp.sip.tls.

enable

sip

added

dir.local.readonly

pres.idleSoftKeys

[MACADDRESS]-directory.xml is not

uploaded.

If set to “0”, appearance of presence

idle soft keys is disabled.

2.2 Version 2.2.1 (Limited Release)

2.2.1 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

2.2.2 Removed Features

None.

Page 4

Release Notes - SIP Application

Changes

2.2.3 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

• 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

2.2.4 Configuration File Parameter Changes

.cfg File

sip

Action

added

Parameter

sec.srtp.offer.HMAC_SHA1_80

Description

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.

2.3 Version 2.2.0

2.3.1 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)

Page 5

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 6

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

• 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

• 36531: Updated all translation dictionary files to rename "Services" menu

entry to "Applications"

2.3.2 Removed Features

• 36079: Removed support for the SoundPoint IP 300 and 500 phones

2.3.3 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

• 26110: Volume level cannot be changed in audio diagnostics mode

• 26231: ACD login failure should cause busy tone to be played

Page 7

Release Notes - SIP Application

Changes

• 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 crash 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 crash 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

• 31994: Trying to delete a null unicode character in the contact list causes the

phone to crash

Page 8

Release Notes - SIP Application

Changes

• 32179: When SAS-VP provisioning is used, the boot server password is visible

in the application log file

• 32816: Phone may crash on subsequent call if using NTLM and received

transfer from a non-NTLM phone

• 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 crash 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

• 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 crash when an 8^thwatched buddy is added

Page 9

Release Notes - SIP Application

Changes

• 35490: SoundPoint IP 320 and 330 phones do not display SAS-VP failure

messages during boot-up

• 36031: If a phone is configured to use TLS for the 2^ndline and TCP for the 1^st,

the 2^ndline 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 crash

• 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

2.3.4 Configuration File Parameter Changes

.cfg File

sip

Action

added

Parameter

voIpProt.SIP.csta

Description

Not currently used, will be used in a

future release.

sip

added

voIpProt.SIP.serverFeatureControl.d See Administrator’s Guide for SIP

nd 2.2.0 for details

voIpProt.SIP.serverFeatureControl.c See Administrator’s Guide for SIP

2.2.0 for details

up.toneControl.bass

Not currently used, will be used in a

future release.

up.toneControl.treble

up.audioSetup.auxInput

up.audioSetup.auxOutput

up.idleTimeout

Not currently used, will be used in a

future release.

Not currently used, will be used in a

future release.

Not currently used, will be used in a

future release.

See Administrator’s Guide for SIP

2.2.0 for details

se.pat.ringer.12.inst.5.type="branch"

se.pat.ringer.12.inst.5.value="-4"

voice.txPacketFilter

sip

added

See Administrator’s Guide for SIP

2.2.0 for details

Page 10

Release Notes - SIP Application

Changes

.cfg File

sip

Action

added

Parameter

voice.codecPref.IP_7000.xxx

Description

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

Several gain and other voice

parameters have been added.

Not currently used, will be used in a

future release.

sip

added

The entire gain section in sip.cfg must

be updated. Failure to do this will

affect the audio performance of the

phone.

Not currently used, will be used in a

future release.

voice.rxEq.hf.IP_7000.xxx

voice.txEq.hf.IP_7000

call.dialtoneTimeOut

sip

Sip

added

See Administrator’s Guide for SIP

2.2.0 for details

call.disableAutoResumeCentralConf Not currently used, will be used in a

erence

call.singleKeyPressConference

future release.

Not currently used, will be used in a

future release.

See Administrator’s Guide for SIP

2.2.0 for details

Not currently used, will be used in a

future release.

Not currently used, will be used in a

future release.

See Administrator’s Guide for SIP

2.2.0 for details

Not currently used, will be used in a

future release.

call.transfer.blindPreferred

call.cellPhoneAutoBridging

bitmap.IP_7000.xxx

log.level.change.srtp

log.level.change.clink

log.level.change.pnetm

log.level.change.peer

Page 11

Release Notes - SIP Application

Changes

.cfg File

Sip

Action

added

Parameter

Description

See Technical Bulletin 25751 for

details.

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

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

license.polling.time

sip

added

See Administrator’s Guide for SIP

2.2.0 for details

Not currently used, will be used in a

future release.

feature.16.name

feature.16.enabled

mb.main.idleTimeout

sip

added

See Administrator’s Guide for SIP

2.2.0 for details

See Administrator’s Guide for SIP

2.2.0 for details

Not currently used, will be used in a

future release.

mb.main.statusbar

pnet.role

Page 12

Release Notes - SIP Application

Changes

.cfg File

sip

Action

changed

Parameter

tone.chord.ringer.46.offDur="200" to

Description

“0”

tone.chord.ringer.46.repeat="2" to

“1”

sip

changed

se.pat.ringer.12.inst.1.type="silence" Note: also added

to “chord”

se.pat.ringer.12.inst.1.value="100"

to “46”

se.pat.ringer.12.inst.5.type=”branch”

and se.pat.ringer.12.inst.5.value="-4"

se.pat.ringer.12.inst.2.type="chord"

to “silence”

se.pat.ringer.12.inst.2.value="46" to

“200”

se.pat.ringer.12.inst.3.type="silence"

to “chord”

se.pat.ringer.12.inst.3.value="2000"

to “46”

se.pat.ringer.12.inst.4.type="branch"

to “silence”

se.pat.ringer.12.inst.4.value="-2" to

“2000”

sip

changed

voice.audioProfile.G722.jitterBufferS Audio performance tuning.

hrink="500" to “1500”

voice.audioProfile.G722.jitterBufferM

ax="160" to “200”

sip

changed

Several gain and other voice

parameters have been changed.

The entire gain section in sip.cfg must

be updated. Failure to do this will

affect the audio performance of the

phone.

voice.rxEq.hd.IP_650.preFilter.enabl Audio performance tuning.

e="1" to “0”

voice.txEq.hs.IP_650.preFilter.enabl

e="1" to “0”

voice.txEq.hd.IP_650.preFilter.enabl

e="1" to “0”

voice.txEq.hf.IP_650.preFilter.enabl

e="1" to “0”

sip

changed

voice.handset.txag.adjust.IP_430="2 Audio performance tuning.

4" to “9”

voice.handset.sidetone.adjust.IP_43

0="-13" to “0”

Multiple parameters in the

ind.anim.xxx, ind.class.xxx and

ind.gi.xxx sections.

The entire indicator section in sip.cfg

must be updated. Failure to do this

will affect the appearance of the

display.

sip

changed

removed

res.finder.minFree=”1200” to “600”

ind.anim.xxx parameters from

CTX_CUSTOM1 to CTX_CUSTOM8

and CTX_UNASSIGNED for all

platforms

These parameters were not used.

sip

removed

added

usb.enable

These parameters were not used.

usb.bulkDrive.enable

usb.bulkDrive.name

reg.x.csta

phone1

Not currently used, will be used in a

future release.

Page 13

Release Notes - SIP Application

Changes

.cfg File

Action

Parameter

Description

See Administrator’s Guide for SIP

2.2.0 for details

phone1

added

reg.x.serverFeatureControl.dnd

reg.x.serverFeatureControl.cf

call.missedCallTracking.x.enabled

phone1

added

See Administrator’s Guide for SIP

2.2.0 for details

See Administrator’s Guide for SIP

2.2.0 for details

See Administrator’s Guide for SIP

2.2.0 for details

phone1

call.callWaiting.ring

000000000000

LICENSE_DIRECTORY

APP_FILE_PATH_SPIP300="sip_21 These are samples of the new fields

2.ld"

which can specify application images

and configuration files for specific

hardware platforms, in this case the

SoundPoint IP 300.

CONFIG_FILES_SPIP300="phone1

_212.cfg, sip_212.cfg”

See Administrator’s Guide for SIP

2.2.0 for details

000000000000

added

APP_FILE_PATH_SPIP500="sip_21 These are samples of the new fields

2.ld"

which can specify application images

and configuration files for specific

hardware platforms, in this case the

SoundPoint IP 500.

CONFIG_FILES_SPIP500="phone1

_212.cfg, sip_212.cfg"

See Administrator’s Guide for SIP

2.2.0 for details

2.4 Version 2.1.2

2.4.1 Added or Changed Features

• 35361: Added ability for parameters in <Ethernet address>.cfg to be

overridden by model- or platform-specific versions

• 35969: Changed behavior of the select button or right arrow button in call lists

and contact directory on SoundPoint IP 320 and 330 to give contact

information instead of acting the same as the dial key

• 36538: Added configurable failover behavior for authentication signaling to

specify that the phone first retries a SIP transaction with the server that has

just sent a 401 or 407 response

Uses new parameters voIpProt.SIP.authOptimizedInFailover and/or

reg.x.auth.optimizedInFailover

• 36647: Added configurable option allowing message waiting indicator to be

displayed although voicemail cannot be accessed

Uses new parameter up.mwiVisible

• 36681: Added logging of version information for configuration files

2.4.2 Removed Features

None.

2.4.3 Corrections

• 34899: Phone may continuously reboot if a configuration change is made then

power is disconnected and the provisioning server is unavailable

Page 14

Release Notes - SIP Application

Changes

• 35873: Registration expiry period is limited to 65535 seconds

• 35914: Scheduled logging stops after 99 days

• 35961: Cannot use call/group/directed pickup on SoundPoint IP 320 and 330

phone while a call is incoming or the phone is off hook

• 35974: SoundPoint IP 320 and 330 phones do not show status for watched

contacts until after the next reboot

• 35979: SoundPoint IP 320 and 330 phones reboot while trying to use call

pickup on a remote hold BLA call

• 36011: After changing termination while in a local conference, the first time the

volume is adjusted the volume slider shows minimum

• 36044: Downloadable character sets are not working correctly in certain

scenarios

• 36053: On SoundPoint IP 320 and 330 phones, Add and Delete soft keys

should not be available in buddy list if roaming buddy feature is disabled

• 36072: On SoundPoint IP 320 and 330 phones, the digit map is not applied to

numbers selected from a call list when in the dial-tone state

• 36074: On SoundPoint IP 320 and 330 phones, the digit map is not correctly

applied when using hot dialing from the second line key

• 36225: Phone may reboot if several voicemail NOTIFY messages are received

from the server in a short interval

• 36233: Specially crafted Via: header in an INVITE can crash the phone

• 36504: A call is dropped if a blind transfer to an invalid number is attempted

• 36581: SoundPoint IP 320 and 330 phones cannot send #nn codes

• 36753: One phone drops the call when 2nd party attempts another blind

transfer to an invalid number

• 36877: All microbrowser text, regardless of which tag is used (except for

"href"), is dim on SoundPoint IP 550 and 650 phones

2.4.4 Configuration File Parameter Changes

.cfg File

sip

Action

added

Parameter

Description

voIpProt.SIP.authOptimizedInFail This parameter controls failover

over

behavior during authentication

signaling.

0 = default behavior which obeys the

RFC

1 = optimization enabled, phone first

retries a SIP transaction with the

server that has just sent a 401 or 407

response

Page 15

Release Notes - SIP Application

Changes

.cfg File

sip

Action

added

Parameter

up.mwiVisible

Description

0 = same behavior as SIP 2.1.1, this

is the default behavior

1 = if msg.mwi.x.callBackMode

parameter is set to “disabled”,

message waiting indicator is

displayed but voicemail cannot be

accessed

sip

changed

added

Changed file header from

$Revision: $ $Date: $

$RCSfile: sip.cfg,v $ $Revision: $

reg.x.auth.optimizedInFailover

This is required to support the new

feature 36681 described above.

phone1

If this parameter is set, it overrides

the global

voIpProt.SIP.authOptimizedInFailover

parameter.

x is the registration index.

See the description for

voIpProt.SIP.authOptimizedInFailover

This is required to support the new

feature 36681 described above.

phone1

changed

Changed file header from

$Revision: $ $Date: $

$RCSfile: phone1.cfg,v $

$Revision: $

Changed file header from

$Revision: $ $Date: $

$RCSfile: 000000000000.cfg,v $

$Revision: $

000000000000

This is required to support the new

feature 36681 described above.

000000000000-

directory~.xml

Changed file header from

$Revision: $ $Date: $

This is required to support the new

feature 36681 described above.

$RCSfile: 000000000000-

directory~.xml,v $ $Revision: $

2.5 Version 2.1.1 C

2.5.1 Added or Changed Features

• 32146: Added support for SoundPoint IP 330

• 33391: Added support for SoundPoint IP 320

• 35415: Added translations for new phrases needed for SoundPoint IP 320 and

330 phones

2.5.2 Removed Features

None.

2.5.3 Corrections

The following issues have been resolved with this release:

Page 16

Release Notes - SIP Application

Changes

35913: SoundPoint IP430, 550, 650 phones may reboot while in a call under certain

network conditions

2.5.4 Configuration File Parameter Changes

None.

2.6 Version 2.1.1

2.6.1 Added or Changed Features

• 33263: Added support for G.729 Annex B SDP signalling per RFC 3555

Note: New parameter voice.vad.signalAnnexB has been added to support this

• 35268: Added support for 16 levels of gray on the LCD of SoundPoint IP 550

and 650 phones

• 35643: Added support for new SoundPoint IP 320 and 330 phones in the

configuration files to allow easier addition of these phones in a future software

release

2.6.2 Removed Features

None.

2.6.3 Corrections

The following issues have been resolved with this release:

• 32273: Failure of call park action results in a dropped call

• 32609: Heavy call volume may cause phone to reject calls due to resource

depletion

• 33390, 35392, 35482: Voice activity detection (VAD) comfort noise generation

(CNG) packets can be discarded by the jitter buffer or interpreted as out-of-

order packets which may result in delayed receive audio when the G.729B

codec is in use

• 33586: The To URI is used in a refer-to header instead of the contact URI

Note: New parameter voIpProt.SIP.useContactInReferTo has been added to sip.cfg

to control the source of the URI used in the refer-to header

• 33647: The phone may reboot because it detects a suspended task even

though that task may have been suspended intentionally

• 33967: An error message is logged if a daylight savings time (DST) start or

stop time of 0 (12am) is selected (although the selection is correctly used)

• 34325: Microbrowser display is closed when shared line is opened on other

phone

• 34431: When changing the configuration of a phone via the web interface, the

phone may lock up

Page 17

Release Notes - SIP Application

Changes

• 34443: A remote-on-hold call on a line is not picked up by the first press of the

line key with some SIP servers

• 34508: In a G.729 call, SoundPoint IP 50X and 60X phones may reboot with a

DSP assertion failure. This problem is more likely in conference calls and can

be reliably reproduced within 20 minutes of the call start.

• 34723: RTCP transmission interval is not consistent with industry norms

• 34772: The value of the DLSR field in RTCP sent by the phone can be wrong by

up to about one second

• 34827: There are two places to configure the microbrowser from the phone

web server

• 34882: The configuration page on the phone web server has two “Event 2”

entries in the Global Log Level Limit drop-down list

• 34906: NOTIFY request without dialog content (an 'empty' NOTIFY request,

such as you would get with a subscription renewal when the line is idle) does

not extinguish LED’s lit as a result of previous active dialogs

• 35049: DSP load graph on SoundPoint IP 550 shows slightly incorrect value

• 35228: Phone may have one-way audio when SDP is received with c line below

m line

• 35293: Soft keys have some missing pixels on the SoundPoint IP 430 when the

microbrowser is accessed

• 35308: A known problem in the SoundPoint IP 430 processor may cause the

phone to reboot with a DSP assertion failure instead of restarting the affected

driver

• 35477: When handset AEC is enabled on SoundPoint IP 50X and 60X phones,

echo may occur on speaker phone when switching between handset and

speaker phone

• 35533: The phone’s web server shows the DST start and stop days as Monday

by default instead of Sunday

• 35537: A saturated transmit signal may cause SoundPoint IP 430 phone to

reboot

• 35573: After selecting the Russian language and accessing the microbrowser,

the phone may freeze

• 36012: Conference host may indicate phone is muted but audio is heard by far

end after one leg ends call

Page 18

Release Notes - SIP Application

Changes

2.6.4 Configuration File Parameter Changes

.cfg

File

sip

Action Parameter

Description

added

voIpProt.SIP.useContactInReferTo

0 = default behavior which is the same as

previous behavior, use URI from initial

call’s To header in REFER’s refer-to

header

1 = use URI from initial call’s Contact

header in REFER’s refer-to header when

setting up a transfer

sip

added

voice.gain.rx.analog.chassis.IP_330

voice.gain.rx.analog.ringer.IP_330

voice.gain.rx.digital.chassis.IP_330

voice.gain.rx.digital.ringer.IP_330

voice.gain.tx.analog.chassis.IP_330

voice.gain.tx.digital.chassis.IP_330

voice.rxEq.hs.IP_330.preFilter.enable

voice.rxEq.hs.IP_330.postFilter.enable

voice.rxEq.hd.IP_330.preFilter.enable

voice.rxEq.hd.IP_330.postFilter.enable

voice.rxEq.hf.IP_330.preFilter.enable

voice.rxEq.hf.IP_330.postFilter.enable

voice.txEq.hs.IP_330.preFilter.enable

voice.txEq.hs.IP_330.postFilter.enable

voice.txEq.hd.IP_330.preFilter.enable

voice.txEq.hd.IP_330.postFilter.enable

voice.txEq.hf.IP_330.preFilter.enable

voice.txEq.hf.IP_330.postFilter.enable

voice.vad.signalAnnexB

New parameters to support SoundPoint

IP 320 and 330 platforms which will be

supported in a future software release. Do

not change these values.

sip

added

A new line can be added to SDP

depending on the setting of this

parameter and the voice.vadEnable

parameter.

Default behavior is the same as

voice.vad.signalAnnexB = 0:

No change to the SDP

voice.vad.signalAnnexB = 1:

If voice.vadEnable=1, add attribute line

a=fmtp:18 annexb=”yes”

below a=rtpmap… attribute line (where

‘18’ could be replaced by another

payload)

If voice.vadEnable=0, add attribute line

a=fmtp:18 annexb=”no”

below a=rtpmap… attribute line (where

‘18’ could be replaced by another

payload)

Page 19

Release Notes - SIP Application

Changes

.cfg

File

sip

Action Parameter

Description

added

voice.handset.rxag.adjust.IP_330

New parameters to support SoundPoint

IP 320 and 330 platforms which will be

supported in a future software release. Do

not change these values.

voice.handset.txag.adjust.IP_330

voice.handset.sidetone.adjust.IP_330

voice.headset.rxag.adjust.IP_330

voice.headset.txag.adjust.IP_330

voice.headset.sidetone.adjust.IP_330

dir.search.field

font.IP_330.1.name

bitmap.IP_330.1.name to

bitmap.IP_330.66.name

ind.idleDisplay.mode

ind.anim.IP_330.38.frame.1.bitmap

ind.anim.IP_330.38.frame.1.duration

ind.gi.IP_330.1.index to

ind.gi.IP_330.10.index

ind.gi.IP_330.1.class to

ind.gi.IP_330.10.class

ind.gi.IP_330.1.physX to

ind.gi.IP_330.10.physX

ind.gi.IP_330.1.physY to

ind.gi.IP_330.10.physY

ind.gi.IP_330.1.physW to

ind.gi.IP_330.10.physW

ind.gi.IP_330.1.physH to

ind.gi.IP_330.10.physH

2.7 Version 2.1.0

2.7.1 Added or Changed Features

• 5844: Enhanced support for server fall-back configurations

• 7275: Microbrowser should auto-navigate to first selectable item

• 7444: Added table support to microbrowser

• 8452: Added microbrowser support to the SoundStation IP 4000

• 9268: Added unique prompt for billing code entry

• 9649: Enhanced '+' global prefix character for E.164 user parts in sip: URIs

• 11572: Added ability to strip or insert leading digits for outgoing calls

• 13497: Updated default daylight savings time rules

• 13818: Added ability to disable message waiting indication on a line by line

basis

• 13882: Added support for setting RTP streams to inactive when on hold

• 14485: Increased maximum number of digit map segments to 30

• 14733: Improved text entry efficiency in the microbrowser

• 14740: Improved visibility of cursor in text entry fields of microbrowser

Page 20

Release Notes - SIP Application

Changes

• 14759: Added microbrowser support to the SoundPoint IP 501 platform

• 14760: Added microbrowser support to the SoundPoint IP 430 platform

• 14900: Changed line-seize subscription failure handling to be biased towards

providing dial tone

• 15934: Added more low end dynamic range to volume control

• 16110: Added support for SoundPoint IP 550 platform

• 16515: Improved "aresDnsLookup: time out on socket select" log message

• 16527: Added a debugging command to display cached DNS NAPTR records

• 17124: Added support for SYSLOG reporting of system status and errors

• 18434: Changed call timer clock display to have no leading colon

• 18966: Added support for adding phone serial number (Ethernet address) to

user agent string in HTTP GET’s used by microbrowser, and modified format

of user agent string used during provisioning process and used by

microbrowser

Example showing format of user agent in HTTP GET’s previously:

User-Agent: Polycom-Microbrowser/1.0 (SIP/2.0.2.0060; SoundPoint IP

PolycomSoundPointIP-SPIP_650) libcurl/7.12.1\r\n

Example showing format of user agent in HTTP GET’s now (with security

sec.tagSerialNo set to 1):

User-Agent: Microbrowser/1.1 PolycomSoundPointIP-SPIP_430-UA/2.1.0.2643

(SN:0004f210013a)

• 19111: Added TCPOnly as a transport option

• 19425: Added microbrowser support for form input elements with checked =

“true” attribute

• 19443: Added microbrowser support for forms within tables

• 19572: Added configurable sticky line seize behavior only for on-hook dialing

2.7.2 Removed Features

None.

2.7.3 Corrections

The following issues have been resolved with this release:

• 7301: Phone doesn't ring if one line has Do Not Disturb enabled

• 16354: Inconsistent error message given when attempting to make a call on an

unregistered line using different methods when call.enableOnNotRegistered is

set to ‘0’

• 16477: When phone is configured for NAPTR transport but server does not

contain NAPTR and SRV, the phone may do SRV lookups for A records or A

lookups for SRV records

Page 21

Release Notes - SIP Application

Changes

• 16899: Phone can send a malformed target URI in some NOTIFY messages in

certain scenario

• 17179: Transfer may fail in some scenarios if the Transfer softkey is pressed

before the second party answers

• 17318: Phone does not update presence status (e.g. to offline) when reboot

initiated

• 17422: When using a bridged line, if a call is transferred to an invalid number it

cannot be retrieved

• 17614: Setting the phone’s own status through "MyStat" does not work

properly

• 17868: Boot server password is displayed in Configuration menu if boot server

is specified as a full URL including user name and password

• 17911: Per-registration DND does not work on SoundPoint IP 430

• 17918: call.enableOnNotRegistered parameter is not working correctly

• 17920: Incorrect icon displayed for offline status when using peer-to-peer

presence

• 18078: When using an LCS server, contacts cannot be added on the phone

when the contact list is empty

• 18147: Expansion modules may display solid background if SoundPoint IP 601

or 650 has maximum number of registrations configured and maximum

number of roaming buddies enabled

• 18198: Value of reg.x.callsPerLineKey parameter is not taken into account

when additional calls are placed using hot (static) dialing

• 18297: VAD/CNG Rx synthesis not working on SoundPoint IP 650

• 18333: Received data on any socket resets timeout of all sockets

• 18393: DTMF levels 3dB lower than configured level when RFC 2833 disabled

• 18501: Incoming call is sent to wrong line in some scenarios when the phone

has an active call and reg.x.lineKeys > 1

• 18688: Value of reg.1.callsPerLineKey parameter is not taken into account

when two lines are configured and reg.2.callsPerLineKey is set to default and

there is a call on hold on both lines

• 18772: SoundPoint IP 650 phone does not show ‘HD’ animation when a wide-

band call is transferred to it

• 18773: After a transfer, a SoundPoint IP 650 phone may incorrectly display the

‘HD’ animation when the call is no longer a wide-band call

• 18785: After receiving a transferred call which is not a wide-band call, a

SoundPoint IP 650 phone may incorrectly display the ‘HD’ animation

• 18985: The log render level for the “sip” module cannot be changed

• 19113: Phone sends incorrect authorization header in some hold scenarios

Page 22

Release Notes - SIP Application

Changes

• 19124: Setting codec preferences using web interface does not work correctly

for SoundPoint IP 650

• 19252: Phone does not send a final NOTIFY to initiator of transfer if the phone

cancels the transfer before it completes

• 19292: SoundPoint IP 650 phone may freeze after restarting after configuration

changed using one of the menus

• 19427: Phone can display “Cache bounced” error message when submitting

forms from the microbrowser

• 19524: Problems resuming a call which is on hold on a remote bridged line for

a specific SIP server

• 19605: Phone may continue to send INVITE’s in specific scenario if a call is

initiated then ended but the SIP servers are not reachable

• 19664: Phone may reboot in some scenarios with log file showing a Null

pointer in a specific task

• 19702: Receipt of a re-transmitted invalid SIP ACK message may cause phone

to reboot

• 19754: Do Not Disturb key cannot be remapped to Null

• 19827: Phone using Bridged Line Appearance can send corrupt message

header in SUBSCRIBE message

• 19875: Phone should use NTP time to check validity of SSL server certificate

• 19876: Phone will lose some memory if microbrowser displays “Cache

bounced” error message due to unresponsive server

• 19883: Handset sidetone level is 3dB too hot on SoundPoint IP 430

• 35063: Power levels reported via CDP for SoundPoint IP 650 are too low

• 35068: Power levels reported via CDP for SoundPoint IP 601 with EM Power

option enabled are too high

2.7.4 Configuration File Parameter Changes

.cfg

Action Parameter

Description

File

phone1

added

reg.x.server.y.lcs

Refer to Technical Bulletin 5844.

Refer to Technical Bulletin 11572.

dialplan.x.applyToUserSend="1"

dialplan.x.applyToUserDial="1"

dialplan.x.applyToCallListDial="0"

dialplan.x.applyToDirectoryDial="0"

reg.x.server.y.transport and

phone1

added

Added “TCPOnly” as a possible value for

these existing parameters.

reg.x.outboundProxy.transport

changed

msg.mwi.x.callBackMode="disabled" to

msg.mwi.x.callBackMode="registration"

(for x = 2, 3, 4, 5, 6) [changed for bug

13818]

sip

added

voIpProt.server.1.lcs

Refer to Technical Bulletin 5844.

Page 23

Release Notes - SIP Application

Changes

.cfg

File

sip

Action Parameter

Description

added

voIpProt.SIP.useSendonlyHold

Can be set to 0 or 1. Null default is 0.

Default in sip.cfg is 1.

If set to 1, the phone will send a reinvite

with a stream mode attribute of “sendonly”

when a call is put on hold. This is the

same as the previous behavior.

If set to 0, the phone will send a reinvite

with a stream mode attribute of “inactive”

when a call is put on hold.

Note:

The phone will ignore the value of this

parameter if set to 1 when the parameter

voIpProt.SIP.useRFC2543hold

is also set to 1 (default is 0).

sip

added

dialplan.applyToUserSend="1"

dialplan.applyToUserDial="1"

dialplan.applyToCallListDial="0"

dialplan.applyToDirectoryDial="0"

dialplan.digitmap.timeOut="3" to

"3|3|3|3|3|3"

tcpIpApp.sntp.daylightSavings.start.mo

nth="4" to “3”

tcpIpApp.sntp.daylightSavings.start.dat

e="1" to “8”

tcpIpApp.sntp.daylightSavings.stop.mon

th="10" to “11”

tcpIpApp.sntp.daylightSavings.stop.day

OfWeek.lastInMonth="1" to “0”

call.stickyAutoLineSeize.onHookDialing Refer to Administrator’s Guide Addendum

for SIP 2.1.

voice.gain.rx.digital.chassis.IP_650="-9" Gain changes required to match new

Refer to Technical Bulletin 11572.

sip

changed

added

Refer to Technical Bulletin 11572.

Changes to support new daylight savings

time rules.

changed

added

to “6”

software load.

voice.gain.rx.digital.ringer.IP_650="-21"

to “-12”

voice.handset.sidetone.adjust.IP_430="

-12" to “-13”

voIpProt.server.x.transport and

voIpProt.SIP.outboundProxy.transport

Added “TCPOnly” as a possible value for

these existing parameters.

2.8 Version 2.0.3 B

2.8.1 Added or Changed Features

• 14874: Added support for SoundPoint IP 650 platform

• 15775: Added support for LCD backlight on SoundPoint IP 650

• 15852: Added support for 32 MB of memory on SoundPoint IP 650

• 15853: Added support for G.722 audio code on SoundPoint IP 650

• 16335: Added support for 8 MB of flash on SoundPoint IP 650

Page 24

Release Notes - SIP Application

Changes

• 16686: Added support for USB diagnostics

• 17132: Added visual indication of wideband audio

2.8.2 Removed Features

None.

2.8.3 Corrections

The following issues have been resolved with this release:

None.

2.8.4 Configuration File Parameter Changes

None.

2.9 Version 2.0.3

2.9.1 Added or Changed Features

None

2.9.2 Removed Features

None.

2.9.3 Corrections

The following issues have been resolved with this release:

• 17981: DHCP initialization incorrect for SoundStation IP 4000 which may cause

boot time problems on some servers

• 18491: Network load reported by SoundPoint IP 430 phones is affected by

traffic which is not destined for the phone

• 18692: Presence subscribe has “application/pidf+xml” in Accept header

although it is not fully supported

• 18766: Ethernet transmit level is low on SoundPoint IP 430 phone

• 18790: Some shared line scenarios do not work with Broadsoft R14 and R13

MP13 releases

• 18919, 11981, 18997: Time stamp in RTCP packets is incorrect

• 19016: SDP containing two “a=” lines causes transfer from a private line to a

shared line to fail

• 19082: Phone seizes wrong line making outbound call to FAC *55

• 19210: Too many messages are logged when “so” is set to level 2

Page 25

Release Notes - SIP Application

Changes

2.9.4 Configuration File Parameter Changes

The following configuration file changes have been included in this build in preparation for

future inclusion of the IP 650 platform in a software release. Support for the IP 650 is not

currently included in this release.

.cfg

File

sip

Action Parameter

Description

added

up.backlight.onIntensity

This parameter controls the intensity of the

LCD backlight when it turns on during

normal use of the phone.

Possible values are 0, 1, 2 or 3.

0 = off

1 to 3 = low, medium, high

Null default is 3 (high).

This parameter controls the intensity of the

LCD backlight when the phone is idle

Possible values are 0, 1, 2 or 3.

0 = off

sip

up.backlight.idleIntensity

1 to 3 = low, medium, high

Null default is 1 (low).

Note: If idleIntensity is set higher than

onIntensity, it will be replaced with the

onIntensity value.

sip

added

voice.codecPref.IP_650.G711Mu

voice.codecPref.IP_650.G711A

voice.codecPref.IP_650.G729AB

voice.codecPref.IP_650.G722

These parameters allow the voice codec

preference list to be set for the SoundPoint

IP 650 phone. By default the G.722 codec is

the first choice.

The use of these parameters is the same as

other voice.codecPref parameters.

These parameters configure the G.722

voice codec. The use of them is the same

as the other voice.audioProfile parameters.

sip

added

voice.audioProfile.G722.payloadSize

voice.audioProfile.G722.jitterBufferMin

voice.gain.rx.analog.chassis.IP_650

voice.gain.rx.analog.ringer.IP_650

voice.gain.rx.digital.chassis.IP_650

voice.gain.rx.digital.ringer.IP_650

These parameters control gain settings

which are specific to the SoundPoint IP 650

phone. The values should not be modified.

voice.gain.tx.analog.chassis.IP_650

voice.gain.tx.digital.chassis.IP_650

voice.rxEq.hs.IP_650.preFilter.enable

voice.rxEq.hs.IP_650.postFilter.enable

voice.rxEq.hd.IP_650.preFilter.enable

voice.rxEq.hd.IP_650.postFilter.enable

voice.rxEq.hf.IP_650.preFilter.enable

voice.rxEq.hf.IP_650.postFilter.enable

voice.txEq.hs.IP_650.preFilter.enable

voice.txEq.hs.IP_650.postFilter.enable

voice.txEq.hd.IP_650.preFilter.enable

voice.txEq.hd.IP_650.postFilter.enable

voice.txEq.hf.IP_650.preFilter.enable

voice.txEq.hf.IP_650.postFilter.enable

sip

added

These parameters control equalization

settings which are specific to the

SoundPoint IP 650 phone. The values

should not be modified.

Page 26

Release Notes - SIP Application

Changes

.cfg

File

sip

Action Parameter

Description

added

voice.handset.rxag.adjust.IP_650

These parameters control gain settings

which are specific to the SoundPoint IP 650

phone. The values should not be modified.

voice.handset.txag.adjust.IP_650

voice.handset.sidetone.adjust.IP_650

voice.headset.rxag.adjust.IP_650

voice.headset.txag.adjust.IP_650

voice.headset.sidetone.adjust.IP_650

dir.local.volatile.8meg

sip

added

This parameter applies only to platforms

with 8 Mbytes of flash memory.

It can be set to 0 or 1 and is 0 by default.

If set to 1, use volatile storage for phone-

resident copy of the directory to allow for

larger size.

dir.local.nonVolatile.maxSize.8meg

This parameter applies only to platforms

with 8 Mbytes of flash memory.

It can be set from 1 to 100. The units are

Kbytes and the default is 100.

This is the maximum size of non-volatile

storage that the directory will be permitted

to consume.

sip

added

log.level.change.usb

This parameter is used to set the logging

detail level for the “usb” module.

prov.fileSystem.ffs0.8meg.minFreeSpac The minimum free space in Kbytes to

reserve in the file system when

downloading files from the boot server.

It is recommended that this value should not

be modified.

The allowed range for this parameter is 5 to

512 and the default is 512.

sip

added

usb.enable

This parameter enables or disables the

USB port on the phone. It can be set to 0 or

1. The Null default is 0.

This parameter enables or disables support

for a USB bulk drive (“memory stick”)

connected to the USB port on the phone. It

can be set to 0 or 1. The Null default is 0.

This parameter is a string which specifies

the name of the mounted USB drive. The

Null default is “usbDrive”.

For the SoundPoint IP 650 platform only,

the values specified by these parameters

are replaced internally with double the

value. This is because the SoundPoint IP

650 platform has 32 Mbytes of memory

instead of 16 Mbytes.

usb.bulkDrive.enable

sip

added

usb.bulkDrive.name

changed

dir.local.volatile.maxSize

prov.fileSystem.rfs0.minFreeSpace

ramdisk.bytesPerBlock

res.finder.sizeLimit

res.finder.minFree

res.quotas.x.value

mb.limits.nodes

mb.limits.cache

2.10 Version 2.0.2

2.10.1 Added or Changed Features

• 8428: Split call signaling processing from "lamp management" processing

Page 27

Release Notes - SIP Application

Changes

• 18356: Emergency routing is not supported on shared lines

2.10.2 Removed Features

None.

2.10.3 Corrections

The following issues have been resolved with this release:

• 6527: Shared line does not ring if incoming call arrives when phone is playing

dial tone then subsequently hangs up

• 8542: Phone does not display second call appearance in specific bridged line

scenario

• 8547: Local ringback is not played if far end does blind transfer without going

on hold

• 15671: Pressing a line key of a shared line when a call is remote-busy ends the

call

• 16662: Shared line can not establish a call if there are two simultaneous

incoming calls

• 18435: If two INVITE’s come close together with SDP containing "a=ptime",

the phone will crash

• 18471: Setting NAT IP address causes truncation or corruption of IP address

in VIA

• 18747: INVITE failover does not work

2.10.4 Configuration File Parameter Changes

None.

2.11 Version 2.0.1 B

2.11.1 Added or Changed Features

None.

2.11.2 Removed Features

None.

2.11.3 Corrections

The following issues have been resolved with this release:

• 18358: Malformed RTCP packets can crash Cisco gateways.

2.11.4 Configuration File Parameter Changes

None.

Page 28

Release Notes - SIP Application

Changes

2.12 Version 2.0.1

The 2.0.1 Release includes all the changes and corrections from Releases 1.6.6 and 1.6.7

2.12.1 Added or Changed Features

• 8072: Added Nortel MCP NAT traversal parameters to config files

• 11678: Added template support in master configuration file

• 16399: Changed behavior when there is an incoming call on a phone – idle dial

digits are no longer cleared when an incoming call is received

• 16645: Added support for NAT keep-alive

• 17412: Added ability to set Ethernet link mode to SoundPoint IP 430

• 17413: Added ability to set Ethernet link mode to SoundStation IP 4000

2.12.2 Removed Features

•

14275: call.callWaiting.prompt has no effect

This parameter has been removed from the configuration files because it is no

longer used.

2.12.3 Corrections

The following issues have been resolved with this release:

• 7723: Name of net logging module is sometimes corrupted in log file

• 12337: Display of SoundPoint IP 430 flickers under fluorescent lights and may

be shifted vertically by a few pixels

• 12382: The phone will freeze if the DNS server address is all zeroes and the

phone uses a FQDN server name

• 12647: Feature keys cannot be reconfigured to perform other functions

• 12749: Phone locks up during CERT PROTOS testing

• 15138: Text in line labels on SoundPoint IP 430 should be moved one pixel left

• 15227: Phone model of SoundPoint IP 430 is incorrect in CDP packets

• 15311: Contrast adjustment range on the SoundPoint IP 430 is unsuitable

• 15729: Phone does not retry connecting to boot server in specific scenario

• 15731: Phone should use Office Communicator model to update LCS presence

status when multiple endpoints share same registration

• 15812: Phone doesn't handle simultaneous 200/OK and CANCEL race

condition

• 16069: When using Russian dictionary, phone reboots after exiting the DHCP

• 16073: Phone does not clear indicators if BLF removed on server

Page 29

Release Notes - SIP Application

Changes

• 16311: Phone with maximum number of line keys configured may have its line

key labels overwritten by roaming buddy records

• 16373: Local conference host cannot end conference if one leg is put on hold

by far end

• 16562: Expansion Module may reboot if the Do Not Disturb key on the phone is

pressed multiple times while the Expansion Module is booting up

• 16577: Local conference host cannot end conference if first leg was put on

hold by far end when conference was created

• 16659: To: and Refer-to: domains incorrect during failover

• 16681: In some scenarios a phone may initiate a call using TCP but send an

ACK using UDP

• 16768: Inconsistent backlight behavior on SoundStation IP 4000 when

resuming a call or conference

• 16904: Excessive logging from “soem” module at boot time in some scenarios

involving Expansion Module

• 17009: Non-numeric characters or an invalid IP address when dialing by IP

may cause the phone to reboot

• 17068: If the silent ringer is selected, an incoming call can only be answered in

hands free mode

• 17102: SoundPoint IP 430 phone locks up instead of rebooting after detecting

an operating system suspended task [bug 17037]

• 17188: “Time” information in placed call list contains incorrect data after a

transfer has been done

• 17257: Phone loses audio when there is an active call on headset and another

incoming call is rejected

• 17206: Local conference host cannot end conference if both legs are put on

hold by far ends

• 17242: Local conference host's state changes to “held” when second leg

holds and invalid soft keys are displayed

• 17271: Phone will not accept a call with a codec with a dynamic payload

identifier

• 17308: Phone displays "In a meeting" status as "Away" when using LCS

server

• 17362: Add or edit directory (speed dial) contact crashes phone when

configured for roaming buddies

• 17370: Phone may reboot if LCS server is used and presence is enabled

without having roaming buddies enabled

Note: If the LCS server is used, the roaming buddies parameter should be enabled

• 17457: Phone may display incorrect soft keys if a digit is pressed then Menu,

Directories or Messages is selected then de-selected

Page 30

Release Notes - SIP Application

Changes

• 17573: In some scenarios, phone sends 603-Decline after 2 rings on SCA line

• 17639: Expansion Module updates should be continuously done in the

background

• 17656: Phone does not handle outbound fragmented packets that are tagged

for VLAN

• 17706: Phone may freeze after regaining connection with LCS server

• 17783: PRACK message goes directly between phones instead of via LCS

server because of no record-route

• 17797: In some scenarios, phone sets its own presence status to 'Away' when

using the LCS server

• 17831: In some scenarios, phone adds itself to its own buddy list when using

the LCS server

• 17976: NTLM signature should include full "From:" URI

2.12.4 Configuration File Parameter Changes

.cfg

File

sip

Action Parameter

Description

removed

call.callWaiting.prompt

sec.srtp.offer, sec.srtp.require,

sec.srtp.key.lifetime

sip

added

voIpProt.SIP.pingInterval

This parameter is used together with

reg.x.proxyRequire. It specifies the number

of seconds between PING messages sent

by the phone.

Default = 0 = disabled.

Possible range is 0 to 3600.

Note: Server support is required before this

feature can be used.

sip

added

res.finder.minFree

This parameter is used to ensure that the

phone will not download resources which

could leave it with insufficient memory to

function correctly. A resource will not be

downloaded if the phone has less memory

free than res.finder.minFree [kBytes].

This parameter can have the values 1 to

2048. The recommended configuration file

value is 1200. If the parameter is left empty

the default is 800.

Notes:

Setting this value too small may affect

functionality of the phone.

Setting this value too large may mean that

some resources are not downloaded at boot

time.

Page 31

Release Notes - SIP Application

Changes

.cfg

Action Parameter

Description

File

phone1

added

reg.x.proxyRequire

This parameter is used together with

voIpProt.SIP.pingInterval. It specifies the

string which is put in the "Proxy-Require"

header.

Default is an empty string which means no

"Proxy-Require" will be sent.

Note: Server support is required before this

feature can be used.

phone1

added

nat.keepalive.interval

This parameter is used to set the interval in

seconds at which phones will send a keep-

alive packet to the gateway/NAT device to

keep the communication port open so that

NAT can continue to function as set up

initially.

Default value is 0 which means the feature

is disabled.

The allowable range is 0 to 3600.

2.13 Version 2.0.0 (Beta Release Only)

Note: The 2.0.0 Release does not include the changes and corrections from SIP releases

1.6.6 and 1.6.7

2.13.1 Added or Changed Features

• 2236: Added support for TLS protocol

• 2307: When the phone reboots due to a fatal error, it should first log any useful

information

• 5403: Added support for the NTLM authentication protocol

• 5404: Added support for Microsoft Live Communications Server authentication

schemes

• 8817: Added support for BLF SCA mode

• 9110: Added support for platform-specific override strings in dictionaries to

allow abbreviated strings for certain platforms

• 9734: Added option to select which registration to use for "presence"

signaling

• 11646: Added IP QoS support for DSCP (DiffServ)

• 11785: Added support for multiple redundant provisioning servers

• 12270: SIP re-registration interval is now configurable

• 12419: Added support for Broadsoft attendant console/BLF feature

• 12426: Added support for peer-to-peer calls using Microsoft Live

Communications Server 2005

Page 32

Release Notes - SIP Application

Changes

• 12427: Added support for calling to and from Windows Messenger 5.1 and

Office Communicator using Microsoft Live Communications Server 2005

• 12938: Added caching of the state of the message-waiting indicator LED

across controlled reboots

• 13038: Changed “DNS Lookup” name to “Transport” in SIP Configuration

menu and on web interface to match parameter name in sip.cfg

• 13080: Added new consultative transfer behavior so that transfer automatically

completes when originator hangs up

• 13100: Added support for individual configuration of secondary dial tone

• 13315: Increased the maximum number of buddies to 8 for all platforms except

SoundPoint IP 600 and 601 which can watch 48 buddies

• 13317: Increased speed dial menu size limit to 99 for all platforms

• 13463: Added IM support with Office Communicator and Windows Messenger

5.1 in Microsoft Live Communications Server 2005 context

• 13509: Added support for reg.x.address configuration parameter to contain

host part

• 13552: Improved boot-up logging

• 13613: Improved support for multiple m lines in SDP

• 13813: Added the ability for file transfers to attempt to contact multiple IP

addresses per DNS name

• 13893: Re-enabled idle micro browser configuration

• 14029: Lowered CPU load associated with RTP processing

• 14209: Added support for getting buddy lists from Microsoft Live

Communications Server 2005

• 14322: Added per-registration "lcs" parameters

• 14323: Added per-registration outbound proxy parameters

• 14348: Added support for connection reuse draft

• 14496: Added presence support with Windows Messenger 5.1 / Office

Communicator in Microsoft Live Communications Server 2005 context

• 14498: Added Windows Messenger 5.1 / Office Communicator-compatible

presence and IM support in peer-to-peer mode

• 14556: Added support for roaming access control lists

• 14610: Added ability to store resource files listed in MISC_FILES field in

<Ethernet Address>.cfg in flash file system. For example a dictionary file can

be listed which should be used if the phone reboots when the boot server is

unavailable.

• 14628: Added support for populating the speed dial list from a roaming

buddies list sent by a Microsoft Live Communications Server 2005

Page 33

Release Notes - SIP Application

Changes

• 14638: Changed source port for TCP/TLS connection to be a random value

above 32766 after each reboot

• 15180: Added configurable maximum number of servers for redundant boot

server feature (11785)

• 15363: Changed call timer format

• 15644: Added a configuration parameter to choose the name of "pval" field in

Dialog

• 15987: Reduced default resource quota limits for tones

• 16047: Added configurable ms-forking support and reject IM when it is enabled

2.13.2 Removed Features

•

12109: Removed configuration parameters for localized call progress tones

In order to still use this feature, see details in 3.1 Upgrading.

•

13447: Removed presence and IM support for Windows Messenger 4.6, 4.7 and

5.0

12350: Removed compiled-in Polycom idle display indicator bitmap

2.13.3 Corrections

The following issues have been resolved with this release:

• 6078: Cannot adjust the volume of the reorder tone when attempting to seize a

shared line which is remotely active

• 7084: Transducer indicator is not cleared after blind transfer on some

platforms

• 9292: IP 4000 reboots upon downloading a wave file with a path containing ‘\’

instead of ‘/’

• 9709: RTCP not sent or received when calls are on hold

• 9815: SoundStation IP 4000 cannot change language after already changing

language 10 to 12 times

• 11177: Fast-Busy sound effect sequencing wrong in specific scenario when

call on hold

• 11588: The local contact directory feature cannot be disabled

• 11952: If destination phone rejects a blind transferred call, the far end does not

hear a busy tone

• 12020: Bridged line with multiple line keys may have one line indicator left in

the remote active state if a peer bridged line hosts a centralized conference

• 12043: Label of CPU Load graph does not change when DSP load is displayed

• 12106: Address of boot server is truncated in Configuration menu on

SoundPoint IP 500 and 501 phones when it exceeds a certain length

Page 34

Release Notes - SIP Application

Changes

• 12155: SoundPoint IP 300 and 301 phones have no “Exit” soft key during the

ACD login process

• 12308: Cannot place a call from the second line on the phone if the first line is

a shared unregistered line

• 12492: SoundPoint IP 601 phone with Expansion Module(s) attached may fail

to load the selected language after rebooting

• 12630: When a shared line is being used on another phone, pressing the line

key for that line can cause the display to show “Enter number” briefly

• 12711: Phone should play default ring tone if Alert-Info URL is invalid

• 12952: There is no way to reset the user password back to the factory default

password

• 13230: No audio on calls resumed from hold in some multiple call scenarios

• 13253: An unregistered SoundStation IP 4000 may reboot if an invalid number

is dialed

• 13320: When the micro browser fetches SSL data this can interrupt audio

transmitted by the phone

• 13358: My Status menu has two “offline” entries

• 13477: Pressing Hold/Resume soft key twice quickly results in three effective

state changes

• 13500: Phone does not use FTP password stored in flash when

OVERRIDES_DIRECTORY and CONTACTS_DIRECTORY are configured in this

format: "FTP://usr@IP/directory"

• 13512: Parsing of URLs in configuration files does not work for some

categories of URLs

• 13579: SDP parser applies wrong logic

• 13793: cnonce generated by the phone is not random

• 13933: Directory menu display is not perfectly cleaned up after deleting all

contacts

• 14069: Phone may behave incorrectly if an incoming call is answered on a

shared line when another phone sharing the line has Do Not Disturb enabled

• 14083: Wrong expire time might be used when there are multiple contact

header lines

• 14126: If a call is placed to a phone with an unread IM, the message-waiting

indicator LED stops flashing

• 14172: Phone will reboot when a contact is added to the contact directory

which already contains over 40 contacts which are being watched

• 14390: Changing the DNS server configuration via the phone’s menu does not

have any effect

• 14400: Phone can take up to 30 minutes to boot when there are TCP timeouts

Page 35

Release Notes - SIP Application

Changes

• 14408: Soft key labels do not get updated correctly after hot dial attempt when

remote shared line is busy

• 14467: If a URL in <Ethernet Address>.cfg specifies a protocol and user name

but no password, the password in flash is not used

• 14635: No welcome sound effect is played on SoundStation IP 4000 phone

• 14664: SoundPoint IP 301 and 501 and SoundStation IP 4000 phones fail

during a reboot if 12 SAS-VP appearances are configured

• 14781: Cannot use special characters for filenames with TFTP boot server

• 14844: A failed download of a pre-existing file causes that file to be deleted

• 14858: Phone reboots if idle micro browser is running and the Status –

Platform - Application menu is displayed

• 15007: If the server address flash parameter is a URL which specifies a

protocol and user name but not password, the password in flash is not used

• 15101: Provisioning of phone stalled forever in specific scenario

• 15145: SAS-VP feature does not work correctly when the filename parameter is

empty

• 15154: Phone does not behave correctly when it is disconnected from the

network and is using SAS-VP

• 15185: Editing problems exist with long strings

• 15214: Headset memory indicator is not restored after adjusting volume on

some platforms

• 15269: When tcpIpApp.sntp.gmtOffset.overrideDHCP is set but no override

value is given, the DHCP based offset is not applied

• 15351: Blind transfer does not drop unless server sends signaling to drop the

call on the originator’s phone. Problem will occur in pure proxy scenarios

only.

• 15368: Character appears to be deleted during editing

• 15412: TFTP URL of configuration file name in log file may be truncated

• 15455: Phone should not reboot if parameters are missing from flash file

system

• 15463: Phone's presence status is not displayed on UI on SoundPoint IP 300

and 301 phones

• 15554: Problems with password entry for very long passwords

• 15561: Phone may reboot after entering a long incorrect password

• 15571: Phone cannot recover in several scenarios involving transferring mixed

URL and E.164 calls

• 15603: The ‘sip:’ field name which appears when using IP dialing should not

be deletable

Page 36

Release Notes - SIP Application

Changes

• 15679: Ring Type 12 (Ringback-style) sounds incomplete after the first ring

• 15694: Phone crashes and reboots when 'Exit' is pressed from Network

Configuration menu in Korean Language

• 15730: If a menu is displayed when a call is missed on the SoundPoint IP 300

and 301 phones, the missed call count is not updated on the idle display

• 15766: Display is incorrect after selecting name dialing then entering and

exiting a call list while dial tone is playing

• 15781: After putting a local conference on hold then splitting the calls then

joining them, the first call may remain on hold

• 15855: In the Instant Msg menu of the SoundPoint IP 300 and 301 phones,

"x/Ascii" is not displayed after pressing the "1/A/a" softkey

2.13.4 Configuration File Parameter Changes

.cfg

File

sip

Action Parameter

Description

added

voIpProt.server.x.expires.overlap

The number of seconds before the

expiration time returned by server ‘x’ at

which the phone should try to re-register.

The phone will try to re-register at half the

expiration time returned by the server if that

value is less than the configured overlap

value.

Default = 60. Minimum = 5, maximum =

65535.

sip

added

voIpProt.SIP.ms-forking

Default = 0. Can be 0 or 1.

0 = Support for MS-forking is disabled.

1 = Support for MS-forking is enabled and

the phone will reject all Instant Message

INVITEs. This parameter is relevant for LCS

server installations.

Note that if any endpoint registered to the

same account has MS-forking

disabled, all other endpoints default back to

non-forking mode. Windows Messenger

does not use MS-forking so be aware of this

behavior if one of the endpoints is Windows

Messenger.

sip

added

voIpProt.SIP.dialog.usePvalue

Default = 0. Can be 0 or 1.

0 = Phone uses “pval” field name in Dialog.

This obeys the draft-ietf-sipping-dialog-

package-06.txt draft.

1 = Phone uses a field name of “pvalue”.

sip

added

voIpProt.SIP.connectionReuse.useAli Default = 0. Can be 0 or 1.

0 = old behaviour

1 = Phone uses the connection reuse draft

which introduces "alias".

se.pat.callProg.15.name="secondary

dial"

Same configuration method as primary dial

tone. Allows a different tone to be

se.pat.callProg.15.inst.1.type="chord" configured for secondary dial tone.

se.pat.callProg.15.inst.1.value="1"

Page 37

Release Notes - SIP Application

Changes

.cfg

File

sip

Action Parameter

Description

added

qos.ip.rtp.dscp

This parameter allows the DSCP of packets

to be specified. If set to a value this will

override the other qos.ip.rtp… parameters.

Default is Null which means the other

qos.ip.rtp… parameters will be used.

Possible values are 0 to 63, EF, AF11,

AF12, AF13, AF21, AF22, AF23, AF31,

AF32, AF33, AF41, AF42 or AF43.

This parameter allows the DSCP of packets

to be specified. If set to a value this will

override the other qos.ip.callControl…

parameters. Default is Null which means the

other qos.ip.callControl… parameters will be

used.

sip

added

qos.ip.callControl.dscp

Possible values are 0 to 63, EF, AF11,

AF12, AF13, AF21, AF22, AF23, AF31,

AF32, AF33, AF41, AF42 or AF43.

Default = 1. Can be 1, 2, 3, …. Must be a

valid line/registration number. If the number

is not a valid line/registration number, it is

ignored.

Specifies the line/registration number used

to send SUBSCRIBE for presence.

mb.idleDisplay.home can be empty or any

fully formed valid HTTP URL. Length up to

255 characters.

sip

added

pres.reg

mb.idleDisplay.home

Default is empty.

This specifies the URL used for the

microBrowser idle display home page.

Example: http://www.example.com/

xhtml/frontpage.cgi?page=home.

If empty, there will be no micro Browser idle

display feature.

sip

added

mb.idleDisplay.refresh

Can be 0 or an integer greater than 5.

Values from 1 to 4 will be ignored, and 5 will

be used instead.

Default = 0

This specifies the period in seconds

between refreshes of the microBrowser idle

display content.

0 = the idle display microBrowser is not

refreshed.

Note: If an HTTP Refresh header is

detected, it will be respected, even if this

parameter is set to 0. The use of this

parameter in combination with the Refresh

HTTP header may cause the idle display to

refresh at unexpected times.

sip

removed

voIpProt.SIP.WM50

For selecting between Windows Messenger

4.7 and 5.0 (no longer supported).

Page 38

Release Notes - SIP Application

Changes

.cfg

File

sip

Action Parameter

Description

removed

lcl.ml.lang.cpt.x,

Removed the parameters used to configure

the call progress tone localization menu.

In order to still use this feature, the old

configuration parameters should be added

to the sip.cfg file and a new parameter,

feature.cpt.enabled, must be added and set

to 1.

lcl.cpt,

lcl.cpt.menu.x,

lcl.cpt.chord.cp.x.y.freq.z,

feature.10.name = cpt-settings

feature.10.enabled = 1

sip

changed

tone.chord.ringer.46.offDur from 200

to 0,

Changes to make ring type 12 work as

expected.

tone.chord.ringer.46.repeat from 1 to

Settings for se.pat.ringer.12

sip

changed

voice.gain.tx.digital.chassis.IP_430

from -3 to 0

voice.handset.txag.adjust.IP_430

from 24 to 21

bitmap.IP_400.61.name from

IdleDefault to “”

Gain corrections for SoundPoint IP 430

platform.

Removed compiled-in Polycom idle display

indicator bitmap.

bitmap.IP_500.61.name from

IdleDefault to “”

bitmap.IP_600.65.name from

IdleDefault to “”

bitmap.IP_4000.66.name from

IdleDefault to “”

sip

changed

HEADSET_MEM IP_300 indicator to

use indicator #50

HEADSET_MEM IP_500 indicator to

use indicator #50

ind.class.4.state.6.index from 48 to 50

ind.anim.IP_400.38.frame.1.bitmap

from IdleDefault to “”

Changed due to rearrangement of other

indicators.

Removed compiled-in Polycom idle display

indicator bitmap.

ind.anim.IP_500.38.frame.1.bitmap

from IdleDefault to “”

ind.anim.IP_500.39.frame.1.bitmap

from IdleDefault to “”

ind.anim.IP_600.38.frame.1.bitmap

from IdleDefault to “”

ind.anim.IP_600.39.frame.1.bitmap

from IdleDefault to “”

ind.anim.IP_4000.38.frame.1.bitmap

from IdleDefault to “”

ind.anim.IP_4000.39.frame.1.bitmap

from IdleDefault to “”

sip

changed

added

res.quotas.1.value from 2000 to 600

Reduced default resource quota limits for

tones.

Default = 0. Can be 0 or 1.

If set to 1 the LCS server is supported for

registration ‘x’.

phone1

reg.x.lcs

phone1

added

reg.x.server.y.expires.overlap

reg.x.outboundProxy.address

Same interpretation as

voIpProt.server.y.expires.overlap for

registration ‘x’.

Same interpretation as

voipProt.SIP.outboundProxy.address for

registration ‘x’.

Page 39

Release Notes - SIP Application

Changes

.cfg

Action Parameter

Description

File

phone1

added

reg.x.outboundProxy.port

Same interpretation as

voipProt.SIP.outboundProxy.port for

registration ‘x’.

Same interpretation as

voipProt.SIP.outboundProxy.transport for

registration ‘x’.

For attendant console / BLF feature. This

specifies the list SIP URI on the server. If

this is just a user part, the URI is

constructed with the server host name/IP

phone1

reg.x.outboundProxy.transport

attendant.uri

phone1

added

attendant.reg

For attendant console / BLF feature. This is

the index of the registration which will be

used to send a SUBSCRIBE to the list SIP

URI specified in attendant.uri. For example,

attendant.reg = 2 means the second

registration will be used.

Specifies the line/registration number which

has roaming buddies support enabled.

Default is empty which means roaming

buddies is disabled. If value < 1 then value

is replaced with 1. This parameter is

relevant for LCS server installations.

Specifies the line/registration number which

has roaming privacy support enabled.

Default is empty which means roaming

privacy is disabled. If value < 1 then value is

replaced with 1. This parameter is relevant

for LCS server installations.

roaming_buddies.reg

roaming_privacy.reg

2.14 Version 1.6.7

2.14.1 Added or Changed Features

• 15930: Added ability to set Ethernet link mode on SoundPoint IP 601

• 15981: Added menu options for setting Ethernet link mode on SoundPoint IP

601

• 16376: Improved response time of phone to SIP messages

• 16482: Added option for phone to be more assertive in negotiating the

preferred codec

• 16500: Added configurable line-seize behavior

2.14.2 Removed Features

None.

2.14.3 Corrections

• 16027: When connecting to voicemail in specific scenario, phone may have no

audio

Page 40

Release Notes - SIP Application

Changes

• 16075: Phone plays re-order tone when taking call off hold in specific scenario

• 16100: BLA line key status is not maintained in specific scenario

• 16116: Cannot register lines 7 to 12 from SIP configuration menu

• 16149: Line key LEDs for BLA lines can switch from one line key to another in

specific scenario

• 16250: Comfort noise received by phone is handled incorrectly

• 16374: Phone keeps sending NOTIFY if 481 received in early NOTIFY

• 16388: Removed DC bias from Tx signal

• 16429: Web interface does not have configuration options for lines 7 to 12

• 16459: Phone is unable to park a call that is received via ACD final destination

• 16480: BLA Led gets stuck and there is a phantom NOTIFY from the phone in a

particular scenario.

• 16485: Notify Talk is ignored if interval between it and 180 is too brief

• 16565: Dialed digits can be lost if they are dialed too quickly after selecting an

SCA line

• 16599: SoundPoint IP 300 and 301 phones reboot when using G.729 codec in a

conference call with SIP 1.6.6 C software

• 16660: Failover to backup SIP server does not occur when hostname of

primary cannot be resolved via DNS

• 16691: Dialog does not get removed after its expiration time in some

scenarios. This addresses #16374 and #16480.

• 16813: Going on and off hook repeatedly on a shared line may result in the line

showing an active call state when the handset is physically on-hook

• 16915: Phone sends SIP requests to port 5060 regardless of

voIpProt.SIP.outboundProxy.port configuration setting

• 17014: When a shared line call is on hold, using on-hook dialing seizes the last

used line instead of the first available line

• 17284: An unnecessary ACK is sent by the phone if no reply is received within

32 seconds

2.14.4 Configuration File Parameter Changes

.cfg Action Parameter

File

Description

sip

added

voIpProt.SDP.answer.useLocalPreferences Can be 0 or 1. Use this new parameter to

have the phone use its own preference

list when deciding which codec to use

rather than the preference list in the offer.

Null default = 0 = disabled.

Page 41

Release Notes - SIP Application

Changes

.cfg Action Parameter

File

Description

sip

added

call.stickyAutoLineSeize

Can be 0 or 1.

Set to 1 to make the phone use "sticky"

line seize behavior. This will help with

features that need a second call object to

work with. The phone will attempt to

initiate a new outgoing call on the same

SIP line that is currently in focus on the

LCD (this was the behavior in SIP 1.6.5).

This may fail due to glare issues in which

case the phone may select a different

available line for the call.

Null default = 0 = disabled (this was the

behavior in SIP 1.6.6).

2.15 Version 1.6.6 C (Limited Distribution)

2.15.1 Added or Changed Features

None.

2.15.2 Removed Features

None.

2.15.3 Corrections

• 16250: Comfort noise received by phone is handled incorrectly. Fixed for

SoundPoint IP 300, 301, 500, 501, 600 and 601 phones.

• 16388: DC bias should be removed from Tx signal on SoundPoint IP 300, 301,

500, 501, 600 and 601 phones

2.15.4 Configuration File Parameter Changes

None.

2.16 Version 1.6.6 B

2.16.1 Added or Changed Features

• Add Support for SoundPoint IP 430 hardware platform

2.16.2 Removed Features

None.

2.16.3 Corrections

None

Page 42

Release Notes - SIP Application

Changes

2.16.4 Configuration File Parameter Changes

.cfg

File

sip

Action Parameter

Description

added

voice.gain.rx.analog.chassis.IP_430,

New gain parameters for SoundPoint IP 430

platform.

voice.gain.rx.analog.ringer.IP_430,

voice.gain.rx.digital.chassis.IP_430,

voice.gain.rx.digital.ringer.IP_430,

voice.gain.tx.analog.chassis.IP_430,

voice.gain.tx.digital.chassis.IP_430,

voice.gain.tx.analog.preamp.chassis.IP

_430

sip

added

voice.rxEq.hs.IP_430.preFilter.enable,

New Rx EQ parameters for SoundPoint IP

voice.rxEq.hs.IP_430.postFilter.enable, 430 platform.

voice.rxEq.hd.IP_430.preFilter.enable,

voice.rxEq.hd.IP_430.postFilter.enable,

voice.rxEq.hf.IP_430.preFilter.enable,

voice.rxEq.hf.IP_430.postFilter.enable

voice.txEq.hs.IP_430.preFilter.enable,

voice.txEq.hs.IP_430.postFilter.enable,

voice.txEq.hd.IP_430.preFilter.enable,

voice.txEq.hd.IP_430.postFilter.enable,

voice.txEq.hf.IP_430.preFilter.enable,

voice.txEq.hf.IP_430.postFilter.enable

voice.handset.rxag.adjust.IP_430,

voice.handset.txag.adjust.IP_430,

voice.handset.sidetone.adjust.IP_430,

voice.headset.rxag.adjust.IP_430,

voice.headset.txag.adjust.IP_430,

voice.headset.sidetone.adjust.IP_430

New Tx EQ parameters for SoundPoint IP

430 platform.

New handset and headset gain adjustments

for SoundPoint IP 430 platform.

sip

added

changed

font.IP_400.1.name

New dynamic font download parameter for

SoundPoint IP 430 platform.

New bitmap parameter for SoundPoint IP

430 platform.

New animation parameters for SoundPoint

IP 430 platform.

Changed the values of some of these

indicator parameters for the SoundPoint IP

430 platform.

bitmap.IP_400.61.name

ind.anim.IP_400.38.frame.1.bitmap,

ind.anim.IP_400.38.frame.1.duration

ind.gi.IP_400…

2.17 Version 1.6.6

2.17.1 Added or Changed Features

• 15491: Added configurable option to enable phone with BLA to send re-INVITE

during conference setup

• 13315: Increased the maximum number of buddies to 8 for all platforms except

SoundPoint IP 600 and 601 which can watch 48 buddies

Page 43

Release Notes - SIP Application

Changes

2.17.2 Removed Features

None.

2.17.3 Corrections

The following issues have been resolved with this release:

• 11658: Phone continues to append to log file on FTP boot server after that file

has reached its configured size limit

• 12613: SoundPoint IP600 and 601 phones may establish a call with no audio

after holding, resuming and ending multiple calls

• 12949: If the phone’s first line is a shared line and cannot obtain dial tone,

pressing the “NewCall” soft key does not activate the first available line

• 14673: Special characters such as ‘@’, ‘:’ and ‘?’ are not accepted as part of

the FTP or HTTP password

• 14968: If the phone reboots, the app.log size can increase past the size limit

• 15002: If the phone’s first line is unregistered, pressing the “NewCall” soft key

does not activate another line

• 15127: Phone may have one-way audio in a call after multiple transfers have

been done

• 15218: If multiple contact header fields contain multiple expire values, the

phone does not always pick the lowest non-zero value

• 15235: Phone will freeze if the SAS-VP server becomes unavailable when the

phone application is starting

• 15339: ACK lacks the same authorization credentials as the INVITE which is a

failure to comply with RFC 3261

• 15419: Blind transfer doesn't work for URL calling

• 15568: A comma in quotes in SIP address headers should be interpreted

correctly

• 15596: Remote phone can force local conference host to resume call

unexpectedly in specific scenario

• 15615: When a shared line call is on hold, lifting the handset seizes the last

used line instead of the first available line

• 14939: Shared line user must press “Answer” soft key twice to answer an

incoming call in some scenarios

• 15907: After a reboot, a phone may show "1 new missed call" which can't be

cleared until another call is missed

• 15982: The SDP session identifier should not be changed on each re-INVITE

• 16021: FTP downloads may fail because incorrect timeouts are used

• 16141: Phone with a shared line loses hot dialed digits when remote shared

line changes state, such as placing an active call on hold

Page 44

Release Notes - SIP Application

Changes

• 16161: Phone with a shared line displays the wrong soft key labels after

attempting to hot dial when the remote shared line is in use

2.17.4 Configuration File Parameter Changes

.cfg

File

sip

Action Parameter

Description

added

call.shared.exposeAutoHolds

call.shared.exposeAutoHolds="1" means

that on a shared line, when setting up a

conference, a re-INVITE will be sent to the

server.

call.shared.exposeAutoHolds="0" means no

re-INVITE will be sent to the server.

Default is “0”.

2.18 Version 1.6.5

2.18.1 Added or Changed Features

• 8072: Added support for Nortel MCP NAT traversal

• 11805: Changed behavior when a local conference is terminated. The remote

conference legs are transferred so that the remote parties can continue the

conversation.

• 13193: Added configuration options to allow configuration file parameters to

override DHCP values for SNTP server address and GMT offset

• 13527: Added support for setting SIP server address from DHCP option 151

• 13509: Added allowing reg.x.address to contain host part instead of being a

user part only

• 13492: CA certificate expiry is no longer checked if SNTP has not been

configured

• 14052: Added flash parameter for SoundPoint IP 601phones to toggle power

requirements in CDP between 5W (no Expansion Modules can be connected)

and 12W (three Expansion Modules can be connected) with a default setting of

This “EM Power” flash parameter is accessible when the SIP application is running

under the Network Configuration menu. Note that no Expansion Modules can be

connected to the phone when the “EM Power” parameter is disabled. The default

setting for this parameter is Enabled (i.e. 12W power requirement). In order for the

correct CDP power requirements to be reported at boot time as well, bootROM

version 3.1.3 is required. See Tech Bulletin TB14052 for details on how to use this

feature.

• 14886: Changed power reported via CDP to platform-specific values

In order for these CDP power requirements to be reported at boot time as well,

bootROM version 3.1.3 is required.

Page 45

Release Notes - SIP Application

Changes

• 15012: Added a workaround to restart the application on the phone if many

tasks get unrealistic task delays during startup (Outstanding issue 11653)

2.18.2 Removed Features

None.

2.18.3 Corrections

The following issues have been resolved with this release:

• 11264: SoundStation IP 4000 hangs when booting if custom DHCP option 150

of type String is used

• 11302: SoundPoint IP 300 and 301 incorrectly truncate displayed line label if

the reg.x.label field is empty and reg.x.address is longer than 4 characters

• 13904: SoundStation IP 4000 always shows LAN Mode as half-duplex

• 14077: Under certain DNS failover conditions, the phone stops sending DNS

and SIP requests

• 14110: Phone does not reset to using “All Certificates” for CA Certificates

after the user chooses the Reset Device Settings menu option

• 14163: Phone incorrectly updates Placed Calls list with an empty entry after

New Call then End Call are pressed

• 14166: Calls answered on a phone with a shared line are incorrectly logged in

the Received Calls list of another phone sharing that line

• 14474: Phone won't upload all log files to TFTP boot server if

LOG_FILE_DIRECTORY specified in <Ethernet Address>.cfg doesn't exist

• 14509: If the SAS-VP xml response has a blank or missing “contactaddr”

element, the phone does not use the “username” field for the contact address

and may lock up during reboot

• 14510: The “username” field in a SAS-VP xml response is not used as the SIP

• 14557: The SAS-VP key is cleared if the user chooses the Reset Device

Settings menu option

• 14634: Blind transfer fails with certain devices due to NOTIFY behavior

• 14684: Problems with text entry interface in custom certificate installation

display

• 14805: Shared lines behave incorrectly if the line registration contains a '.'

• 14935: Phone begins to ring when there is no incoming call in specific shared

line scenario

• 15104: SoundStation IP 4000 CDP does not advertise new link duplex levels

correctly

• 15122: Time displayed on phone changes from correct to incorrect shortly

after a reboot in some scenarios

Page 46

Release Notes - SIP Application

Changes

• 15162: Phone clears application log file during a warm boot even if the upload

to the boot server failed

2.18.4 Configuration File Parameter Changes

.cfg

File

sip

Action Parameter

Description

added

voIpProt.server.dhcp.available

1 = check with the DHCP server for SIP

server IP address.

0 = do not check with DHCP server.

Default = 0.

sip

added

voIpProt.server.dhcp.option

voIpProt.server.dhcp.type

Option to request from the DHCP server if

voIpProt.server.dhcp.available = 1.

Allowable range is 128 – 255. There is no

default value for this parameter, it must be

filled in with a valid value.

0 = IP address

1 = string

Type to request from the DHCP server if

voIpProt.server.dhcp.available = 1.

There is no default value for this parameter,

it must be filled in with a valid value.

These parameters determine whether

configuration file parameters override DHCP

added

sip

tcpIpApp.sntp.address.overrideDHCP

and

tcpIpApp.sntp.gmtOffset.overrideDHCP parameters for the SNTP server address

and GMT offset. The default is 0 which

means that DHCP values will override

configuration file parameters. A value of 1

means that configuration file parameters will

override DHCP values.

2.19 Version 1.6.4

2.19.1 Added or Changed Features

• 12278: Added support for SAS-VP v3 XML configuration transactions

• 12883: Added sending and processing the “early-only” flag in the “replaces”

header to support RFC 3891 in call pickup

• 12890: Added accepting SDP with telephone-event on the first line

• 13492: Disabled CA certificate expiry checking when SNTP has not been

configured

2.19.2 Removed Features

None.

2.19.3 Corrections

The following issues have been resolved with this release:

• 7707: LED which shows mute and incoming-call and message-waiting status

can show incorrect state

• 8598: There is no "1/A/a" soft key when editing Forward contact

Page 47

Release Notes - SIP Application

Changes

• 12626: Phone reboots on installation of a custom certificate

• 12882: Display of time and date on SoundStation IP 4000 gets truncated during

a call if the line label is 10 digits long

• 13034: Phone should stop sending further NOTIFY messages if 481 response

received

• 13318: SoundStation IP 4000 file system is smaller than it should be

• 13440: Changes in APP_FILE_PATH cause unnecessary application changes

Note: This fix requires bootROM version 3.1.2.

• 13507: The phone at times incorrectly maintains two SUBSCRIBEs for call-info

• 13533: The phone doesn’t upload directory or configuration override files to a

TFTP server unless they already exist on the server

• 13553: The “entity” field in a dialog for private lines can be improperly

formatted

• 13554: A phone in the offering state should send a NOTIFY response to a

dialog SUBSCRIBE request for all lines except Bridged Lines

• 13582: “Supported” header in INVITE should contain “replaces” instead of

“replace”

• 13699: VLAN from CDP may work intermittently on SoundStation IP 4000

• 14116: After a blind transfer fails, the call cannot be retrieved

• 14219: RTP sequence numbering starts at wrong value after a call is resumed

from hold

• 14220: Lost packets statistics are incorrect after far end resumes a call

• 14387: A display name containing a ‘.’ is not displayed in some scenarios

2.19.4 Configuration File Parameter Changes

None.

2.20 Version 1.6.3

2.20.1 Added or Changed Features

• 11358: Added configurable subdirectories for configuration and contact

directory override files

• 12761: Added support for setting flash parameters from configuration file

• 13029: Added support for new dialog event package draft

draft-ietf-sipping-dialog-package-06.txt

• 13030: Added support for new BLA draft

draft-anil-sipping-bla-02.txt

• 13222: Changed maximum number of XML retries for SAS-VP to be equal to 7

days

Page 48

Release Notes - SIP Application

Changes

• 13931: Added notice of file system fix for bug 13361 to header of SoundStation

IP 4000 binary image

2.20.2 Removed Features

• 13025: Disabled url-dialing in main partner configuration files

2.20.3 Corrections

The following issues have been resolved with this release:

• 11271: Phone repeatedly tries to upload log file when log.render.file parameter

disabled

• 12449: Shared line continues to ring after receiving a CANCEL event in some

scenarios

• 12470: Misplaced comma in date display for two possible date formats

• 12748: Caller ID shows IP address when PSTN caller is unknown

Note: The “url-dialing” feature must be disabled in order for the IP address to be

hidden

• 12842: Some characters sent in the dial string should be escaped but are not

• 13089: Outbound proxy port greater than 6535 does not work

• 13198: Long date format gets changed to short date format after first call

• 13223: All user agent headers for SAS-VP v3 must include <Ethernet address>

• 13228: Audio lost for the first call after rejecting the second incoming call if

headset or hands free is used

• 13235: Repeatedly holding and resuming a call can result in no audio when the

call is resumed

• 13258: Frequent registration retry to an inactive server after server failover can

result in the phone being unable to put a call on hold

• 13285: Unverified SSL connections were allowed to SAS-VP server

• 13289: Long date format does not work if a shared line calls itself

• 13361: IP 4000 security certificate (HTTPS and SAS-VP provisioning) can

become corrupt after file system activity.

Note: BootROM must be upgraded to version 3.1.2 as instructed in Technical

Bulletin TB13361

• 13517: Hands free dial-tone volume can become very quiet after significant

volume adjustment

Page 49

Release Notes - SIP Application

Changes

2.20.4 Configuration File Parameter Changes

.cfg File

Action Parameter

Description

000000000000

added

CONTACTS_DIRECTORY,

New fields which can specify a directory on

the boot server in which contact overrides

(<Ethernet address>-directory.xml) and

configuration overrides (<Ethernet

OVERRIDES_DIRECTORY

voIpProt.SIP.dialog.useSDP

feature.9.enabled

address>-phone.cfg) should be stored.

0 or Null: New dialog event package draft is

used (no SDP in dialog body).

1: For backwards compatibility, use this

setting to send SDP in dialog body.

The “url-dialing” feature must be disabled by

setting feature.9.enabled=”0” in order to

prevent unknown callers from being

sip

added

changed

identified on the display by an IP address.

2.21 Version 1.6.2

2.21.1 Added or Changed Features

None.

2.21.2 Removed Features

None.

2.21.3 Corrections

The following issues have been resolved with this release:

• 9580: Changes in <Ethernet address>.cfg will not be detected during

configuration polling

• 11190: Incorrect time zone is used for one to two minutes after a reboot

• 12552: Phone reboots if line keys on Expansion Module are pressed rapidly

and continuously

• 12841: Far end phone continues to ring if near end phone ends call prior to far

end answering in specific shared-line scenario

• 12951: Malformed RTP packets received by phone can cause it to crash

2.21.4 Configuration File Parameter Changes

None.

2.22 Version 1.6.1

2.22.1 Added or Changed Features

• 12296: Pressing and holding unassigned line key adds a directory contact

• 12366: Application log is uploaded shortly after reboot

Page 50

Release Notes - SIP Application

Changes

2.22.2 Removed Features

None.

2.22.3 Corrections

The following issues have been resolved with this release:

• 11388: Phone does not get a CDP response reliably in some scenarios

• 12208: Indicator for watched contact remains red if speed dial line removed

• 12247: Two-stage dialing user interface not correct

• 12348: Handsfree and handset buttons do not work correctly to answer call

when silent ringer is selected

• 12364: Cannot establish a centralized conference from one of the conference

legs

• 12475: One-Touch Voicemail dialing does not support multiple lines correctly

• 12506: INVITE message never tried on backup proxy when primary server fails

over

• 12640: CDP word on SoundPoint IP 601 needs to advertise maximum power to

Cisco switch

• 12775: Phone cannot join more than two legs to centralized conference

2.22.4 Configuration File Parameter Changes

.cfg Action Parameter

File

Description

sip

changed

voice.audioProfile.xxx parameter values and

voice.gain.xxx parameter values

Use the new values for these

parameters.

2.23 Version 1.6.0 (Beta only)

2.23.1 Added or Changed Features

• 4614: Added display of date and time during a call

• 9046: Added support for SoundPoint IP Expansion Module

• 9108, 10480: Added support for SoundPoint IP 601 hardware platform

• 9660: Pressing and holding an assigned speed dial "line key" opens the

contact directory to that entry

• 11540: Improved speed dial key assignment

When perusing the contact directory, pressing and holding an unassigned line key

assigns the in-focus directory entry to that key as a speed dial. A confirmation beep

is heard.

When a new directory entry is added, the speed dial index is automatically assigned

the next available value.

• 11731: Calls from more than one SIP registration (line) can be joined

Page 51

Release Notes - SIP Application

Changes

• 11849: Added support for transfer dispatch during consultation call

proceeding state

New parameter for this is voIpProt.SIP.allowTransferOnProceeding which will

normally not need to be changed.

• 12093: Added a Forward menu so that forwarding can be modified at any time

2.23.2 Removed Features

None.

2.23.3 Corrections

The following issues have been resolved with this release:

• 7521: Transfer from a shared line can be interrupted

• 8507: Directory search does not produce all matches for some last names

• 9790: Outbound proxy transport selection should be clear

New parameter for this is voIpProt.SIP.outboundProxy.transport.

• 9827: A keypad-initiated reboot waits for dial tone to time out before starting

• 11583: Phone does not upload log file when it exceeds render file size

• 11738: Audio Diagnostics don’t work for headset mode

• 11762: Headset indicator/icon can blink during a call between two phones

using the same bridged line which have headset memory enabled

• 11790: Multi-tap entry doesn't work for the very first character entered for URL

dialing

• 11846: 484 response should be treated as an error in ringback state

• 11848: No stuttered dial tone when a line has a message waiting

• 11940: Phone holds the call when a fourth party is added to a centralized

conference

• 11946: Some clock date format selections do not work

• 12032: Pressing headset button in ringing state does not answer call when

headset memory is enabled

• 12066: After editing contact directory items, the “Save” soft key can get

relabeled as “Search”

• 12191: The menu produced when the Directories key is pressed should not

include the “Messages” option

• 12221: ‘-1’ displayed as number of different priority messages for voice

message feature when data is missing

• 12227: Phone attempts to forward a call to a shared line if Auto Divert is

enabled for the contact making the call

• 12247: Two-stage dialing does not work

Page 52

Release Notes - SIP Application

Changes

• 12284: Time handling for DHCP needs to be improved

• 12289: Common audio equalization tables should be grouped together

• 12323: Exiting Display Diagnostics with termination key does not stop display

diagnostics

• 12333: "Direct" and "Group" soft keys can appear when directed and group

call pickup features are disabled

• 12370: Ringing can be heard during a connected call mixed with audio when

there is a high number of unanswered incoming calls

• 12541: Error messages can appear in log file after putting two calls on hold

2.23.4 Configuration File Parameter Changes

.cfg Action Parameter

File

Description

sip

added

voIpProt.SIP.allowTransferOnProceeding

0 = don’t allow transfer during

consultation call proceeding state

1 = do allow it (1 is the default)

Same function and possible values as

existing voIpProt.server.x.transport

parameter. Default is DNSnaptr.

voIpProt.SIP.outboundProxy.transport

voice.gain.rx.analog.chassis.IP_601,

voice.gain.rx.analog.ringer.IP_601,

voice.gain.rx.digital.chassis.IP_601,

voice.gain.rx.digital.ringer.IP_601,

voice.gain.tx.analog.chassis.IP_601,

voice.gain.tx.digital.chassis.IP_601,

voice.gain.tx.analog.preamp.chassis.IP_601

voice.aec.xxx

Gains specifically for the IP 601

platform.

sip

changed

Changed parameter values. Do not

modify these.

Changed parameter values. Do not

modify these.

This whole section has changed and

must be used. Do not modify these.

voice.ns.xxx

added/

removed

added/

removed

added

voice.rxEq.xxx

voice.txEq.xxx

This whole section has changed and

must be used. Do not modify these.

log.level.change.sotet,

log.level.change.ttrs

Added log level control for logging

related to Expansion Module.

Page 53

Release Notes - SIP Application

3. Notes

Notes

3.1 Upgrading

This section lists the changes that should be made to configuration files when using the

centralized (boot server) provisioning model. For general guidelines, see the Updating and

Rebooting information in Section 4.3 of the Administrator Guide.

3.1.1 From Version 2.2.1 to 2.2.2

3.1.1.1 Mandatory Changes

• None.

3.1.1.2 Optional Changes

• TCP Keep-Alive message when using TLS

Configure the tcpIpApp.KeepAlive parameters as detailed in Section 2.1.4 if using

TLS and there is a risk of the TCP connection being improperly terminated.

• Read-only Contact Directory

If it is desired to centrally manage the phones directory, the user can be restricted

from making any changes. To enable this capability set dir.local.read-only = “1”

• Disable Presence (MyStat and Buddies) soft-keys when using the Presence

feature signalling

Some call servers use the phones ‘Presence’ feature for controlling BLF capability

but don’t implement the full suite of Presence options. To avoid giving the user

visibility to this setting, the idle soft-keys may be removed from the phone UI by

setting pres.idleSoftKeys=”0”

3.1.2 From Version 2.2.0 to 2.2.1

3.1.2.1 Mandatory Changes

• None.

3.1.2.2 Optional Changes

• None

3.1.3 From Version 2.1.2 to 2.2.0

3.1.3.1 Mandatory Changes

• New configuration file settings for audio

The entire “voice” section in the latest sip.cfg must be used to ensure good audio

quality.

• New configuration file settings for indicators

The entire “indicators” section in the latest sip.cfg must be used to ensure correct

icons on the display.

Page 54

Release Notes - SIP Application

Notes

3.1.4 From Version 2.1.1 C to 2.1.2

3.1.4.1 Mandatory Changes

• Adding logging of version information for configuration files

In order for this new feature to work, the latest version of all configuration files must

be used.

3.1.4.2 Optional Changes

• Using different versions of configurable items in <Ethernet address>.cfg for

different phone models or platforms

Different phone models or platforms can be configured to use different application

files, configuration files, log file directory etc. See technical bulletin TB35361 for

details.

• Optimizing failover behavior for authentication signaling

Use the new parameters voIpProt.SIP.authOptimizedInFailover in sip.cfg and

reg.x.auth.optimizedInFailover in phone1.cfg to change the phone’s failover behavior

during authentication signaling if desired.

• Viewing message waiting indicators while still retaining one-touch voicemail

access when multiple lines are configured

If a phone has multiple lines with just one registration set to have

msg.mwi.x.callBackMode = “registration” and all others set to have

msg.mwi.x.callBackMode = “disabled” but it is desirable to be able to see message

waiting indicators for all lines and still retain one-touch voicemail access, set the new

parameter up.mwiVisible to 1 in sip.cfg.

3.1.5 From Version 2.1.1 to 2.1.1 C

3.1.5.1 Mandatory Changes

None.

3.1.5.2 Optional Changes

None.

3.1.6 From Version 2.1.0 to 2.1.1

3.1.6.1 Mandatory Changes

None.

3.1.6.2 Optional Changes

• Using URI from call’s contact header in refer-to header

Set the parameter voIpProt.SIP.useContactInReferTo to 1 in sip.cfg if the URI from

the initial call’s Contact header should be used in REFER’s refer-to header when

setting up a transfer. The previous and default behavior is to use the URI from the

initial call’s To header.

• Supporting G.729 Annex B SDP signalling per RFC 3555

If the new parameter voice.vad.signalAnnexB in sip.cfg is set to 1, a new attribute

Page 55

Release Notes - SIP Application

Notes

line will be added to SDP. See details in 2.6.4 Configuration File Parameter

Changes.

3.1.7 From Version 2.0.3 to 2.1.0

3.1.7.1 Mandatory Changes

• Using a Microsoft LCS Server

It may be required to set the new parameters voIpProt.server.x.lcs (in sip.cfg) and

reg.x.server.y.lcs (in phone1.cfg) if the phone registers to a Microsoft LCS server.

3.1.7.2 Optional Changes

• Using “inactive” stream mode attribute when a call is put on hold

The default behavior is for the “sendonly” stream mode attribute to be used when a

call is put on hold. This behavior can be changed to use the “inactive” attribute. In

order to configure this behavior, the parameter voIpProt.SIP.useSendonlyHold must

be set to 0.

• Digit map extension support

The digit map can be configured to remove, add or replace digits. For details see

Technical Bulletin 11572.

• Restricting transport to TCP

The transport used by the phone can be restricted to TCP. This means the phone

will not attempt to fail over to UDP if TCP fails. A new “TCPOnly” option has been

added to all parameters which control the transport used by the phone.

• Adding “sticky line seize” behavior for hot-dial (on-hook) dialing

If sticky behavior is desired for hot dialing this can be configured using the new

call.sticky.AutoLineSeize.onHookDialing parameter. Hot dialing sticky behavior can

be configured to be different than normal new call sticky behavior. “Stickiness” refers

to using the same line for a new call as the last-used line when a call has been put

on hold.

3.1.8 From Version 2.0.3 to 2.0.3 B

3.1.8.1 Mandatory Changes

None.

3.1.8.2 Optional Changes

None.

3.1.9 From Version 2.0.2 to 2.0.3

3.1.9.1 Mandatory Changes

None.

3.1.9.2 Optional Changes

None.

Page 56

Release Notes - SIP Application

Notes

3.1.10 From Version 2.0.1 to 2.0.2

3.1.10.1 Mandatory Changes

None.

3.1.10.2 Optional Changes

None.

3.1.11 From Version 2.0.0 to 2.0.1

3.1.11.1 Mandatory Changes

None.

3.1.11.2 Optional Changes

• Using template support in master configuration file

The master configuration file may contain the string “[MACADDRESS]”. This will be

replaced with the MAC address of the phone. For example, the file

000000000000.cfg may refer to [MACADDRESS]phone.cfg which will be replaced

with something like 0004f2100137phone.cfg. This can make provisioning more

efficient.

• Adding Nortel MCP NAT traversal

The new parameters voIpProt.SIP.pingInterval and reg.x.proxyRequire should be

configured if this feature is needed.

• Adding NAT keepalive

If NAT keepalive is required, the new parameter nat.keepalive.interval should be set

to a non-zero value.

3.1.12 From Version 1.6.7 to 2.0.0

3.1.12.1 Mandatory Changes

• Using the phone’s menu to select call progress tones

This feature has been removed from the default configuration of the phone. In order

to still use this feature, the old configuration parameters should be added to the

sip.cfg file and a new parameter, feature.cpt.enabled, must be added and set to 1.

Old configuration parameters are feature.10.name=”cpt-settings”,

feature.10.enabled=”1”, and the entire localization – multilingual – language –

callProgTones section and the entire localization – callProgTones section.

3.1.12.2 Optional Changes

• Adding IP QoS support for DSCP (DiffServ)

Add the parameters qos.ip.rtp.dscp and qos.ip.callContol.dscp for DSCP. A valid

value is either a number or string as follows

1) Any number from 0 to 63

2) EF

3) Any of AF11, AF12, AF13, AF21, AF22, AF23, AF31, AF32, AF33, AF41, AF42,

AF43

Page 57

Release Notes - SIP Application

Notes

The rules are:

1) When qos.ip.rtp.dscp has a valid value, then it overrides the following:

i) qos.ip.rtp.min_delay

ii) qos.ip.rtp.max_throughput

iii) qos.ip.rtp.max_reliability

iv) qos.ip.rtp.min_cost

v) qos.ip.rtp.precedence

2) Similarly when qos.ip.callControl.dscp has a valid value, then it overrides

qos.ip.callControl.min_delay etc.

3.1.13 From Version 1.6.6 to 1.6.7

3.1.13.1 Mandatory Changes

• Selecting “sticky” line seize behavior

To have the same line seize behavior as SIP 1.6.5, set call.stickyAutoLineSeize to 1

in sip.cfg.

3.1.13.2 Optional Changes

• Overriding codec preferences received from far end

To allow the phone to override the list of codec preferences received by the phone,

set voIpProt.SDP.answer.useLocalPreferences to 1 in sip.cfg.

3.1.14 From Version 1.6.5 to 1.6.6

3.1.14.1 Mandatory Changes

None.

3.1.14.2 Optional Changes

• Sending re-INVITE to server during conference setup on BLA

Set call.shared.exposeAutoHolds to 1 in sip.cfg

3.1.15 From Version 1.6.4 to 1.6.5

3.1.15.1 Mandatory Changes

• None.

3.1.15.2 Optional Changes

• Getting SIP server address from DHCP

The SIP server address can be obtained from a DHCP server if the new parameters

voIpProt.server.dhcp.available, voIpProt.server.dhcp.option and

voIpProt.server.dhcp.type are configured correctly.

• Using configuration file values for SNTP parameters instead of DHCP values

If the configuration file settings for the SNTP server address or GMT offset should be

used instead of the values obtained from a DHCP server, set one or both of the new

parameters tcpIpApp.sntp.address.overrideDHCP and

tcpIpApp.sntp.gmtOffset.overrideDHCP to 1.

Page 58

Release Notes - SIP Application

Notes

• Reducing the power requirements reported via CDP for a SoundPoint IP 601

A new flash parameter “EM Power” is available under the Network Configuration

menu of SoundPoint IP 601 phones. If this is set to “Enabled” the phone will report

power requirements of 12W which is sufficient to power three Expansion Modules. If

the parameter is set to “Disabled” the phone will report power requirements of 5W

and no Expansion Modules can be connected to the phone. By default this

parameter will be set to “Enabled” when the phone is upgraded to 1.6.5. BootROM

version 3.1.3 is required in order for the same power requirements to be reported at

boot time. Please refer to Tech Bulletin TB14052 for details on upgrade/downgrade

process with respect to this parameter.

3.1.16 From Version 1.6.3 to 1.6.4

3.1.16.1 Mandatory Changes

None.

3.1.16.2 Optional Changes

None.

3.1.17 From Version 1.6.2 to 1.6.3

3.1.17.1 Mandatory Changes

• Dialog event package draft backwards compatibility

If the old dialog event package draft behavior is desired (SDP is sent in dialog body),

set the new voIpProt.SIP.dialog.useSDP parameter in sip.cfg to 1.

3.1.17.2 Optional Changes

• Changing the destination of phone-specific override file uploads

Use the new CONTACTS_DIRECTORY and OVERRIDES_DIRECTORY fields in

000000000000.cfg.

• Preventing IP address caller ID display when PSTN caller is unknown

The “url-dialing” feature must be disabled in order for the IP address to be hidden.

3.1.18 From Version 1.6.1 to 1.6.2

3.1.18.1 Mandatory Changes

None

3.1.19 From Version 1.6.0 to 1.6.1

3.1.19.1 Mandatory Changes

• Voice Configuration Parameters Updated

Some parameters in the “voice” section of sip.cfg have been modified and this entire

section is required when using SIP 1.6.1.

Page 59

Release Notes - SIP Application

Notes

3.2 Outstanding Issues

The following issues will be fixed in a subsequent release.

Note: Polycom has switched to a different issue tracking system which has caused the

reference numbers in these release notes to be different to earlier versions. When the

issues are addressed the numbers in this release note can be used to track in which

version the issue is addressed.

• 24398: No Layer 2 QoS support for signaling protocol (TCP)

Workaround: The default QOS parameters will still be used for TCP signaling

packets, and these may be specified in the sip.cfg configuration file. Layer3 QoS

settings are supported.

• 24805: Cannot answer an incoming call while directory is being saved

Workaround: None.

• 26615: Subnet mask forces all packets through gateway when not using DHCP

and when using the wrong subnet mask for the network class in use, for

example using 192.168.X.X addresses with a 255.255.0.0 subnet mask

Workaround: Use the correct subnet mask.

• 26920: Centralized conference fails due to RTP port being slow to open in

some cases

Workaround: None.

• 27469: Local Conferencing on IP4000 phones is disabled if G.729 is in the

Codec preference list

Workaround: Disable G.729 as a Codec option on the phone by setting

voice.codecPref.IP_4000.G729AB=””

• 28508: Phone crashes after receiving high call rate (4 unanswered calls every

18 seconds)

Workaround: Reduce the incoming call rate.

• 29344: HTTP Digest Authentication does not work on IIS

Workaround: Use a different form of authentication, a different protocol or a different

server

• 29946: Log files are not uploaded if an Apache 2.0.X boot server requires

authentication

Workaround: Turn off authentication or use version 1.3.3X of the Apache server.

• 30086: Boot servers running explicit FTPS are not supported

Workaround: Use implicit FTPS or HTTPS.

• 30371: Pattern generator for tones does not work well for the case of a single

repeating chord

Workaround: Start the pattern with a short period of silence then the desired initial

chord. Loop back to the desired initial chord instead of the initial silence.

• 30903: Packet Loss statistics ‘jump’ if calls are transferred.

Workaround: If using the packet loss statistics for troubleshooting purposes make a

note of the Packet Loss value after the transfer and apply a correction based on this

to subsequent calculations.

Page 60

Release Notes - SIP Application

Notes

• 32476: IP601 does not work correctly when Presence feature is enabled with

LCS server without using Roaming Buddies

Workaround: Enable roaming buddies by setting roaming_buddies.reg to the LCS

registration number.

• 32611: BLA line can not place and hold more than 10 calls

Workaround: For BLA lines ensure that call.callsPerLineKey is set to 10 or lower.

• 32816: Phone crashes on subsequent call if using NTLM and received transfer

from non-NTLM phone

Workaround: Ensure that all phones involved in a transfer use NTLM, or do not use

NTLM authentication

• 32994: SoundPoint IP 650 phone may have an incomplete display with only

shades of grey after booting up

Workaround: Cycle power to the phone to make it boot again

• 33063: Active FTP mode is not supported for phone provisioning

Workaround: Configure the ftp server for Passive FTP operation.

• 33445: LCS Presence and dialing from Buddy Lists does not work across

‘Federations’

Workaround: To dial contacts across federations program a speed dial with the SIP

URI of the contact. There is no workaround for watching ‘Federated Buddy’ status

from the phone.

• 33593: Shared line does not show remote active for the second incoming call if

callsPerLineKey parameter is set to 1

Workaround: Set callsPerLineKey parameter to a value greater than 1.

• 34454: If microbrowser is enabled and refreshes are too frequent and pages

contain large images, the phone may crash

Workaround: Do not refresh microbrowser too frequently in configuration settings or

by rapidly pressing the Refresh softkey. Design the pages so that the content is

within reasonable limits.

• 34743: A phone may freeze when it receives a check-sync if the resources on

the phone are heavily used by downloaded wave files or large or complex

microbrowser pages

Workaround: Reduce the RAM disk size configured in sip.cfg (this will reduce the

amount of space available for downloaded wave files and other resources) by setting

ramdisk.nBlocks to 3072. Design web pages used by the microbrowser carefully.

• 36969: SoundStation IP 4000 phone does not display Japanese language

properly.

Workaround: None.

• 37391: The Phone may fail to boot if the contact directory contains improper

XML syntax.

Workaround: Ensure that the contact directory is in a proper XML format.

• 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.

Workaround: Ensure that the server is configured to respond to the REFER that

ends the conference.

Page 61

Release Notes - SIP Application

Reference Documents

• 37391: Brief audio ‘noise’ due to SRTP encryption key change..

Workaround: To minimize the frequency of occurrence configure the

sec.srtp.key.lifetime as long as possible.

•

37437: When SRTP is used with both Authentication and Encryption enabled

on SoundPoint IP 301, 501, 600 and 601 platforms, and three-way conferencing

is enabled the phone will re-boot when a local conference is attempted.

Workaround: Disable local conferencing by setting sec.srtp.leg.allowLocalConf="0"

(this is the default setting) or disable SRTP Authentication. See Technical Bulletin

25751 for details.

•

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.

Workaround: Set allowTransferOnProceeding=”0” which prevents a transfer from

occurring during the proceeding state.

•

39419: Maximum Backlight Intensity setting has very little effect on

SoundPoint IP 560 phones.

Workaround: None.

39490: In some call scenarios the phone may not display the SRTP secure line

icon even though the call is encrypted.

Workaround: None.

Note: The phone does not ever indicate that a call is encrypted when it is not.

39630: Using SoundPoint IP 330/320 phone with LCS2005; Blocking a roaming

buddy from the Privacy list also prevents the user from viewing the 'Blocked'

buddy's status

•

Workaround: Do not block user’s from viewing your status if you wish to view their’s

4. Reference Documents

• Administrator Guide – SoundPoint IP SIP – Version 2.2.0

• Technical Bulletins 5844, 11572 35311, 35361 – may be obtained from the Polycom

web-site Support Knowledge-Base www.polycom.com/support/voip

• Technical Bulletin 25751 is available from the Polycom PRC.

Page 62

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