Polycom SOUNDPOINT IP 430, SOUNDPOINT IP 300, SOUNDPOINT IP 501, SOUNDPOINT IP 600, SOUNDPOINT IP 601 User Manual

Release Notes
SIP Application
SoundPoint® and SoundStation® IP
Version 1.6.7
7 July 2006
Part Number 3804-11530-167
Copyright © 2006 Polycom, Inc. All rights reserved.
Release Notes - SIP Application
Copyright © 2006 Polycom, Inc. All rights reserved.
Release Notes - SIP Application Table of Contents
Table of Contents
1. GENERAL................................................................................................................................... 1
1.1 SYSTEM REQUIREMENTS........................................................................................................ 1
2. CHANGES................................................................................................................................... 1
2.1 VERSION 1.6.7 ....................................................................................................................... 1
2.1.1 Added or Changed Features......................................................................................... 1
2.1.2 Removed Features......................................................................................................... 1
2.1.3 Corrections ................................................................................................................... 1
2.1.4 Configuration File Parameter Changes ....................................................................... 2
2.2 VERSION 1.6.6 C (LIMITED DISTRIBUTION) ........................................................................... 3
2.2.1 Added or Changed Features......................................................................................... 3
2.2.2 Removed Features......................................................................................................... 3
2.2.3 Corrections ................................................................................................................... 3
2.2.4 Configuration File Parameter Changes ....................................................................... 3
2.3 VERSION 1.6.6 B.................................................................................................................... 3
2.3.1 Added or Changed Features......................................................................................... 3
2.3.2 Removed Features......................................................................................................... 3
2.3.3 Corrections ................................................................................................................... 3
2.3.4 Configuration File Parameter Changes ....................................................................... 4
2.4 VERSION 1.6.6 ....................................................................................................................... 4
2.4.1 Added or Changed Features......................................................................................... 4
2.4.2 Removed Features......................................................................................................... 5
2.4.3 Corrections ................................................................................................................... 5
2.4.4 Configuration File Parameter Changes ....................................................................... 6
2.5 VERSION 1.6.5 ....................................................................................................................... 6
2.5.1 Added or Changed Features......................................................................................... 6
2.5.2 Removed Features......................................................................................................... 7
2.5.3 Corrections ................................................................................................................... 7
2.5.4 Configuration File Parameter Changes ....................................................................... 8
2.6 V
ERSION 1.6.4 ....................................................................................................................... 8
2.6.1 Added or Changed Features......................................................................................... 8
2.6.2 Removed Features......................................................................................................... 8
2.6.3 Corrections ................................................................................................................... 8
2.6.4 Configuration File Parameter Changes ....................................................................... 9
2.7 VERSION 1.6.3 ....................................................................................................................... 9
2.7.1 Added or Changed Features......................................................................................... 9
2.7.2 Removed Features....................................................................................................... 10
2.7.3 Corrections ................................................................................................................. 10
2.7.4 Configuration File Parameter Changes ..................................................................... 11
2.8 V
ERSION 1.6.2 ..................................................................................................................... 11
2.8.1 Added or Changed Features....................................................................................... 11
2.8.2 Removed Features....................................................................................................... 11
2.8.3 Corrections ................................................................................................................. 11
2.8.4 Configuration File Parameter Changes ..................................................................... 11
2.9 VERSION 1.6.1 ..................................................................................................................... 11
Copyright © 2006 Polycom, Inc. Page i
Release Notes - SIP Application Table of Contents
2.9.1 Added or Changed Features....................................................................................... 11
2.9.2 Removed Features....................................................................................................... 12
2.9.3 Corrections ................................................................................................................. 12
2.9.4 Configuration File Parameter Changes ..................................................................... 12
2.10 VERSION 1.6.0 ..................................................................................................................... 12
2.10.1 Added or Changed Features....................................................................................... 12
2.10.2 Removed Features....................................................................................................... 13
2.10.3 Corrections ................................................................................................................. 13
2.10.4 Configuration File Parameter Changes ..................................................................... 14
2.11 VERSION 1.5.2 ..................................................................................................................... 14
2.11.1 Added or Changed Features....................................................................................... 14
2.11.2 Removed Features....................................................................................................... 15
2.11.3 Corrections ................................................................................................................. 15
2.11.4 Configuration File Parameter Changes ..................................................................... 16
2.12 VERSION 1.5.1 ..................................................................................................................... 16
2.12.1 Added or Changed Features....................................................................................... 16
2.12.2 Removed Features....................................................................................................... 17
2.12.3 Corrections ................................................................................................................. 17
2.12.4 Configuration File Parameter Changes ..................................................................... 19
3. NOTES.......................................................................................................................................21
3.1 DISTRIBUTION FILES............................................................................................................ 21
3.2 UPGRADING ......................................................................................................................... 22
3.2.1 From Version 1.6.6 to 1.6.7........................................................................................ 22
3.2.2 From Version 1.6.5 to 1.6.6........................................................................................ 22
3.2.3 From Version 1.6.4 to 1.6.5........................................................................................ 23
3.2.4 From Version 1.6.3 to 1.6.4........................................................................................ 23
3.2.5 From Version 1.6.2 to 1.6.3........................................................................................ 23
3.2.6 From Version 1.6.1 to 1.6.2........................................................................................ 24
3.2.7 From Version 1.6.0 to 1.6.1........................................................................................ 24
3.2.8 From Version 1.5.2 to 1.6.0........................................................................................ 24
3.2.9 From Version 1.5.1 to 1.5.2........................................................................................ 24
3.3 O
UTSTANDING ISSUES.......................................................................................................... 25
4. REFERENCE DOCUMENTS................................................................................................. 27
Page ii Copyright © 2005 Polycom, Inc.
Release Notes - SIP Application General
1. General
These release notes apply to version 1.6.7 of the SoundPoint IP SIP application. For more information, refer to the documents listed in Section 4.
1.1 System Requirements
Platform BootROM version
SoundPoint IP 300 2.6.1 or greater SoundPoint IP 301 2.6.1 or greater SoundPoint IP 430 3.1.3 or greater SoundPoint IP 500 2.6.1 or greater SoundPoint IP 501 2.6.1 or greater SoundPoint IP 600 2.6.1 or greater SoundPoint IP 601 3.1.0 or greater SoundStation IP 4000 3.1.2 or greater
2. Changes
2.1 Version 1.6.7
2.1.1 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
2.1.2 Removed Features
None.
2.1.3 Corrections
16027: When connecting to voicemail in specific scenario, phone may have no audio
Copyright © 2006 Polycom, Inc. Page 1
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
2.1.4 Configuration File Parameter Changes
.cfg File
sip added voIpProt.SDP.answer.useLocalPreferences Can be 0 or 1. Use this new parameter to
Action Parameter Description
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 2 Copyright © 2006 Polycom, Inc.
Release Notes - SIP Application Changes
.cfg
Action Parameter Description
File
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).
2.2 Version 1.6.6 C (Limited Distribution)
2.2.1 Added or Changed Features
None.
2.2.2 Removed Features
None.
2.2.3 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
2.2.4 Configuration File Parameter Changes
None.
2.3 Version 1.6.6 B
2.3.1 Added or Changed Features
Add Support for SoundPoint IP 430 hardware platform
2.3.2 Removed Features
None.
2.3.3 Corrections
None
Copyright © 2006 Polycom, Inc. Page 3
Release Notes - SIP Application Changes
2.3.4 Configuration File Parameter Changes
.cfg
Action Parameter Description
File
sip added voice.gain.rx.analog.chassis.IP_430,
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,
voice.rxEq.hs.IP_430.postFilter.enable, 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
sip added 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
sip added 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
sip added font.IP_400.1.name New dynamic font download parameter for sip added bitmap.IP_400.61.name New bitmap parameter for SoundPoint IP sip added ind.anim.IP_400.38.frame.1.bitmap,
ind.anim.IP_400.38.frame.1.duration
sip changed ind.gi.IP_400… Changed the values of some of these
New gain parameters for SoundPoint IP 430 platform.
New Rx EQ parameters for SoundPoint IP 430 platform.
New Tx EQ parameters for SoundPoint IP 430 platform.
New handset and headset gain adjustments for SoundPoint IP 430 platform.
SoundPoint IP 430 platform. 430 platform.
New animation parameters for SoundPoint IP 430 platform.
indicator parameters for the SoundPoint IP 430 platform.
2.4 Version 1.6.6
2.4.1 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 4 Copyright © 2006 Polycom, Inc.
Release Notes - SIP Application Changes
2.4.2 Removed Features
None.
2.4.3 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” softkey 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” softkey
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” softkey 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
Copyright © 2006 Polycom, Inc. Page 5
Release Notes - SIP Application Changes
16161: Phone with a shared line displays the wrong softkey labels after attempting to hot dial when the remote shared line is in use
2.4.4 Configuration File Parameter Changes
.cfg
Action Parameter Description
File
sip 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”.
2.5 Version 1.6.5
2.5.1 Added or Changed Features
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 5W
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.
15012: Added a workaround to restart the application on the phone if many tasks get unrealistic task delays during startup (Outstanding issue 11653)
Page 6 Copyright © 2006 Polycom, Inc.
Release Notes - SIP Application Changes
2.5.2 Removed Features
None.
2.5.3 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
login name for authentication of SIP messages
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
15162: Phone clears application log file during a warm boot even if the upload
to the boot server failed
Copyright © 2006 Polycom, Inc. Page 7
Release Notes - SIP Application Changes
2.5.4 Configuration File Parameter Changes
.cfg
Action Parameter Description
File
sip 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 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.
sip added voIpProt.server.dhcp.type 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.
sip added tcpIpApp.sntp.address.overrideDHCP
and tcpIpApp.sntp.gmtOffset.overrideDHCP
These parameters determine whether configuration file parameters override DHCP 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.
2.6 Version 1.6.4
2.6.1 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
2.6.2 Removed Features
None.
2.6.3 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" softkey when editing Forward contact
12626: Phone reboots on installation of a custom certificate
Page 8 Copyright © 2006 Polycom, Inc.
Release Notes - SIP Application Changes
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
2.6.4 Configuration File Parameter Changes
None.
2.7 Version 1.6.3
2.7.1 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
Copyright © 2006 Polycom, Inc. Page 9
Release Notes - SIP Application Changes
13931: Added notice of file system fix for bug 13361 to header of SoundStation IP 4000 binary image
2.7.2 Removed Features
13025: Disabled url-dialing in main partner configuration files
2.7.3 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 handsfree is used
13235: Repeatly 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 filesystem activity.
Note: BootROM must be upgraded to version 3.1.2 as instructed in Technical Bulletin TB13361
13517: Handsfree dial-tone volume can become very quiet after significant volume adjustment
Page 10 Copyright © 2006 Polycom, Inc.
Release Notes - SIP Application Changes
2.7.4 Configuration File Parameter Changes
.cfg File Action Parameter Description
000000000000 added CONTACTS_DIRECTORY,
OVERRIDES_DIRECTORY
sip added voIpProt.SIP.dialog.useSDP 0 or Null: New dialog event package draft is
sip changed feature.9.enabled The “url-dialing” feature must be disabled by
New fields which can specify a directory on the boot server in which contact overrides (<Ethernet address>-directory.xml) and configuration overrides (<Ethernet address>-phone.cfg) should be stored.
used (no SDP in dialog body). 1: For backwards compatibility, use this
setting to send SDP in dialog body. setting feature.9.enabled=”0” in order to
prevent unknown callers from being identified on the display by an IP address.
2.8 Version 1.6.2
2.8.1 Added or Changed Features
None.
2.8.2 Removed Features
None.
2.8.3 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
2.8.4 Configuration File Parameter Changes
None.
2.9 Version 1.6.1
2.9.1 Added or Changed Features
12296: Pressing and holding unassigned line key adds a directory contact
12366: Application log is uploaded shortly after reboot
Copyright © 2006 Polycom, Inc. Page 11
Release Notes - SIP Application Changes
2.9.2 Removed Features
None.
2.9.3 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
2.9.4 Configuration File Parameter Changes
.cfg File
sip changed voice.audioProfile.xxx parameter values and
Action Parameter Description
Use the new values for these
voice.gain.xxx parameter values
parameters.
2.10 Version 1.6.0
2.10.1 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 12 Copyright © 2006 Polycom, Inc.
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
2.10.2 Removed Features
None.
2.10.3 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
Copyright © 2006 Polycom, Inc. Page 13
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
2.10.4 Configuration File Parameter Changes
.cfg
Action Parameter Description
File
sip added voIpProt.SIP.allowTransferOnProceeding 0 = don’t allow transfer during
consultation call proceeding state 1 = do allow it (1 is the default)
sip added voIpProt.SIP.outboundProxy.transport Same function and possible values as
existing voIpProt.server.x.transport parameter. Default is DNSnaptr.
sip added 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
sip changed voice.aec.xxx Changed parameter values. Do not sip changed voice.ns.xxx Changed parameter values. Do not sip added/
removed
sip added/
removed
sip added log.level.change.sotet,
voice.rxEq.xxx This whole section has changed and
voice.txEq.xxx This whole section has changed and
log.level.change.ttrs
Gains specifically for the IP 601 platform.
modify these. modify these. must be used. Do not modify these.
must be used. Do not modify these. Added log level control for logging
related to Expansion Module.
2.11 Version 1.5.2
2.11.1 Added or Changed Features
11356: Changed configuration of presence and instant messaging features to be disabled by default
11552: Added phone UI and web interface configuration support for lineKeys and callsPerLineKey
Page 14 Copyright © 2006 Polycom, Inc.
Release Notes - SIP Application Changes
2.11.2 Removed Features
11816: Pressing a line key will no longer terminate a call
2.11.3 Corrections
The following issues have been resolved with this release:
9491: Empty "to" header may be sent in some cases
9776: Parsing errors when dealing with the override file
9817: Configuration override file gets unnecessary extra parameters
11189: User can corrupt the directory by editing it when “presence” feature is
disabled
11343: Pressing handsfree or headset button activates handset if handset is
off hook
11409: Provisioning may not work reliably with the proftpd FTP server on
Linux
11417: Phone may not be able to boot from a remote subnet
11426: Secondary dial tone plays incorrectly on certain digit maps
11466, 11558: Provisioning may fail using HTTPS if a custom certificate is
used
11556: Stored authentication key from a SAS-VP server is deleted when the
phone is reset to factory defaults
11558: Provisioning may fail using HTTPS if a custom certificate is used
11575: SoundPoint IP300/301 doesn't give warning message if duplicate IP is
detected by DHCP client
11584: Automatic key repeats do not work
11595: Phone displays URL encoded digits when dialing
11599: Check-sync and polled configuration change features do not work
11600: Phone ignores maximum password length parameters
11608: Disabling "presence" feature does not remove it from phone's menu
11609: Disabling “messaging” feature on SoundStation IP 4000 and
SoundPoint IP30x disables voice message feature as well
11612: When Do Not Disturb per-registration is enabled, the Do Not Disturb
“clear all” soft key is missing
11616: CANCEL requests include tag when they shouldn't
11633: Phone should use flash credentials when boot server URL lacks them
11641: Phone shows an error message on the display when Hold is invoked on
the last available call appearance
11644: Join does not work from the last available call appearance
Copyright © 2006 Polycom, Inc. Page 15
Release Notes - SIP Application Changes
11665: Pressing the headset button in ringing state does not answer call when headset memory is enabled
11685: Line configuration cannot be changed using web server
11739: A call can be lost when Split is used under certain circumstances
11760: Custom certificate gets corrupted if SAS-VP is used
11788: Pressing "New Call" soft key auto dials the previous number entered
using on-hook dialing if the previous call failed
11789: The "more" soft key for establishing a conference can disappear, hiding the “Join” soft key
11798: There is an incompatability when using EPSV with proftpd
2.11.4 Configuration File Parameter Changes
.cfg File Action Parameter Description
sip changed feature.1.enabled,
feature.2.enabled changed from 1
to 0 sip changed voIpProt.server.x.transport Explicitly set default to DNSnaptr phone1 changed reg.x.server.y.transport Explicitly set default to DNSnaptr
Presence and Instant Messaging are disabled by default.
2.12 Version 1.5.1
2.12.1 Added or Changed Features
966: A single call will always show up in the first call appearance position
1509: Improved menu hierarchy
1842: Added visual "status" to contacts assigned to speed dial bins
3924: Added conference feature enhancement to "join" calls in progress
7204: Added flashing time/date until successful SNTP response
7663: Added ability to specify boot server address as URL per RFC 1738
This requires bootROM 3.0 or greater.
7894: Added support for having more than one line key associated with the
same SIP identity
This includes a new feature – pressing and holding down the line key provides call information about a call which is on hold on that line key.
7899: Added support for the application to provision its own configuration files
7900: Added application support for HTTP and HTTPS boot server transport This requires bootROM 3.0 or greater.
For HTTPS, if the time on the phone is wrong the SSL certificate may be rejected. Configure SNTP to obtain an accurate time.
8055: Added support for SAS-VP v2 management This requires bootROM 3.0 or greater.
Page 16 Copyright © 2006 Polycom, Inc.
Release Notes - SIP Application Changes
8521: Added a menu entry to format the file system
8786: Added display of name and number on incoming caller ID
9053: Added support for displaying a useful CID when display name is
uninformative
9096: Added customization options for SSL certificates
9299: Added allowing all files in <MAC>.cfg to be full URL's
9323: Removed requirement for at least two audio codecs to be configured
9496: Merged sip.cfg and ipmid.cfg configuration files into new sip.cfg file
9548: Added allowing user to disable time and date display
9579: Added allowing specific master configuration file to be specified in boot
server URL
9588: Changed offering LED animation to continuous 2 Hz flash, rather than
intermittent
9659: Added feature to split conferences and consultation calls into separate
calls
9675: Added feature to allow conference initiation from call hold context
9694: Changed example directory file to no longer use silent ring type for
contacts
9710: Changed default hold signaling to be the RFC 3261 style
10806: Added build ID to software revision stamps in User-Agent header
11235: Added support for arrow-key call-list shortcuts when phone is playing
dial tone
2.12.2 Removed Features
11973: Removed support for port mode FTP server configurations
Use an FTP server/firewall that supports passive mode connections.
2.12.3 Corrections
The following issues have been resolved with this release:
737: Phone will not accept IP packets bigger than 38,000 bytes
2311: Line labels do not line up with line keys on SoundPoint IP 600
3707: Can't use speed dial when one call already on Hold
7952: FTP transfers should remove partially written files in a failure scenario
8050: Parameters which were not changed are saved in configuration override
file
8333: Improve source data for random device
8416: Bridged Line second call appearance is incorrect in specific scenario
Copyright © 2006 Polycom, Inc. Page 17
Release Notes - SIP Application Changes
8616: Incorrect message on display for incoming call on shared line on SoundPoint IP 4000
8674: Missing remote hold call appearance in specific Bridged Line scenario
8755: For TCP, the response to a request should try the remote port that sends
the request first
8771: IP 4000 cannot download large directory file
8801: Phone ignores X-Syl-Line-ID and mixes call appearances
8873: When a new DHCP lease is obtained, the updated DNS information is not
used
8962: Active Bridged Line cannot switch to incoming call
9090: Clock date menu choice ending in ‘YYYY’ not displayed properly
9135: Random string for CNONCE value for digest authentication should be
limited to the base64 character set
9187: GMT offset and SNTP address set in flash are ignored if parameters exist in configuration file but have no associated value (i.e. are empty)
9243: Web server buttons not labeled, and some labels are incorrect
9326: DST not working for Southern Hemisphere
9452: DTMF tones not recognized by specific IVR after shared line remote
resume
9481: Phone will attempt to download files indefinitely if connection to FTP server lost
9482: Phone waits for an error response from the FTP server when none is forthcoming
9584: Call duration missing from placed call list items on IP 4000
9601: DNS resolution fails when downloading [mac]-phone.cfg
9735: Web interface of SoundStation IP 4000 phone edits some non-IP 4000
parameters
10825: Phone should not collect digits after dial tone has timed out
11285: SIP authentication password stored in [mac]-phone.cfg file
11303: SoundPoint IP 300 phone loses contrast settings during reboot
11348: Large DHCP messages get truncated
11350: SoundStation IP 4000 phone can lock up when a key is pressed
11458: Audio loss on one leg of conference after second conference
automatically put on Hold (first conference is Resumed)
11516: Off-by-one error when ringTypes are saved
11548: Cannot change administrator password or user password on
SoundPoint IP 300
Page 18 Copyright © 2006 Polycom, Inc.
Release Notes - SIP Application Changes
11559: ACD login does not work on SoundStation IP 4000
11563: ACD available/unavailable functions work differently on Bridged and
Private lines
11573: Pressing Handsfree button does not put you back to handset when
handset is off hook
2.12.4 Configuration File Parameter Changes
.cfg File Action Parameter Description
ipmid removed All parameters The contents of this file have been
added to sip.cfg and this file is no longer used.
sip added All “ipmid.cfg” parameters The contents of the old ipmid.cfg file
have been added to sip.cfg.
sip added call.callsPerLineKey The number of calls or conferences
which may be active or on hold per line key on the phone. For the IP 600, range is 1 to 24 and default is 24.
For all other phones, range is 1 to 8 and default is 8.
sip changed voIpProt.SIP.useRFC2543hold Changed the default value to "0" (it
used to be “1”) which means that RFC 3261-style hold signalling is the default.
sip changed voice.gain.rx.analog.chassis.IP300
voice.gain.rx.analog.ringer.IP300
sip changed call.shared.oneTouchResume This applies to SoundStation IP 4000
sip changed ind.gi.IP_600.x… Changed values used for locating line sip removed ind.pattern.8.step.3 to 6 An incoming call causes the LED to
sip removed all .obs parameters from logging
section
phone1 added reg.x.ringType The ring type for each registration can
phone1 added reg.x.lineKeys The number of line keys on the phone
Changed name to voice.gain.rx.analog.chassis.IP_300 voice.gain.rx.analog.ringer.IP_300
phones only in this build. For all other phones, one-touch resume is the default. In order to view call information about a call on hold on another phone with a shared line – press and hold down the line key for a few seconds.
key labels. Update this whole section. flash continuously at 2Hz rather than
flash intermittently. These parameters are no longer
used. be configured. Range is 1 to 22.
Note: ring type number 1 is “silent ring”.
to be associated with registration ‘x’. Range is 1 to the maximum number of line keys on the phone (IP 300 = 2, IP 500 = 3, IP 600 = 6, IP 4000 = 1). Default is 1.
Copyright © 2006 Polycom, Inc. Page 19
Release Notes - SIP Application Changes
.cfg File Action Parameter Description
phone1 added reg.x.callsPerLineKey The number of calls or conferences
which may be active or on hold per line key for a specific registration on the phone. This will override the global call.callsPerLineKey parameter in sip.cfg. Same range and defaults as call.callsPerLineKey above.
000000000000 removed ipmid.cfg from list of
CONFIG_FILES
The ipmid.cfg file is no longer used.
Page 20 Copyright © 2006 Polycom, Inc.
Release Notes - SIP Application Notes
3. Notes
3.1 Distribution Files
The following files constitute the 1.6.7 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 Description
sip.ld
sip.cfg main core and SIP configuration file phone1.cfg example per-phone SIP configuration
000000000000.cfg example master configuration file 000000000000-directory~.xml example per-phone local contact directory XML file (edit and then
SIP application executable, App Version
1.6.7.0094 for SoundPoint IP 430
1.6.7.0098 for all other platforms IP 300 2345-11300-001: 1.6.7 IP 301 2345-11300-010: 1.6.7 IP 430 2345-11402-001: 1.6.7 IP 500 2345-11500-001: 1.6.7 2345-11500-010: 1.6.7 2345-11500-030: 1.6.7 2345-11500-020: 1.6.7 IP 501 2345-11500-040: 1.6.7 IP 600 2345-11600-001: 1.6.7 IP 601 2345-11605-001: 1.6.7 IP 4000 2201-06642-001: 1.6.7
remove ‘~’ from name to seed phones which have no directory)
Copyright © 2006 Polycom, Inc. Page 21
Release Notes - SIP Application Notes
Files Description
SoundPointIP-dictionary.xml dictionary files for multilingual support include (no IP 30X support):
Chinese, China (for IP 60X 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 60X and IP 4000 only) Korean, Korea (for IP 60X and IP 4000 only) Norwegian, Norway Portuguese, Portugal Russian, Russia Spanish, Spain Swedish, Sweden
SoundPointIPWelcome.wav start up welcome sound effect
3.2 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.
3.2.1 From Version 1.6.6 to 1.6.7
3.2.1.1 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.
3.2.1.2 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.
3.2.2 From Version 1.6.5 to 1.6.6
3.2.2.1 Mandatory Changes
None.
3.2.2.2 Optional Changes
Sending re-INVITE to server during conference setup on BLA Set call.shared.exposeAutoHolds to 1 in sip.cfg
Page 22 Copyright © 2006 Polycom, Inc.
Release Notes - SIP Application Notes
3.2.3 From Version 1.6.4 to 1.6.5
3.2.3.1 Mandatory Changes
None.
3.2.3.2 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.
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.
3.2.4 From Version 1.6.3 to 1.6.4
3.2.4.1 Mandatory Changes
None.
3.2.4.2 Optional Changes
None.
3.2.5 From Version 1.6.2 to 1.6.3
3.2.5.1 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.
3.2.5.2 Optional Changes
Changing the destination of phone-specific override file uploads
Use the new CONTACTS_DIRECTORY and OVERRIDES_DIRECTORY fields in
000000000000.cfg.
Copyright © 2006 Polycom, Inc. Page 23
Release Notes - SIP Application Notes
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.
3.2.6 From Version 1.6.1 to 1.6.2
3.2.6.1 Mandatory Changes
None
3.2.7 From Version 1.6.0 to 1.6.1
3.2.7.1 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.
3.2.8 From Version 1.5.2 to 1.6.0
3.2.8.1 Mandatory Changes
Voice Configuration Parameters Updated Many parameters in the “voice” section of sip.cfg have been modified and this entire section is required when using SIP 1.6.0.
Transfer On Proceeding Enabled by Default In SIP 1.5.2 there was no option to complete a transfer during the proceeding state of a consultation call. In SIP 1.6.0 this has been added and it is enabled by default. Set the parameter voIpProt.SIP.allowTransferOnProceeding to 0 if this feature is not wanted.
Selecting the Transport for an Outbound Proxy The transport used by an outbound proxy is determined by the new parameter voIpProt.SIP.outboundProxy.transport. If this parameter is missing, the default of NAPTR will be used. In SIP 1.5.X the outbound proxy transport was determined by the voIpProt.server.1.transport or reg.x.server.1.transport parameters but these are no longer taken into account.
3.2.9 From Version 1.5.1 to 1.5.2
3.2.9.1 Mandatory Changes
Presence and Instant Messaging Disabled by Default These features have been disabled in sip.cfg by setting feature.1.enabled and feature.2.enabled to 0. If these features are required they must be enabled in sip.cfg.
Page 24 Copyright © 2006 Polycom, Inc.
Release Notes - SIP Application Notes
3.3 Outstanding Issues
The following issues will be fixed in a subsequent release.
4310: No 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.
5085: Cannot answer an incoming call while directory is being saved
Workaround: None.
6527: Shared line does not ring if incoming call arrives when phone is playing
dialtone then subsequently hangs up
Workaround: None.
8532: 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.
8547: Local ringback is not played if far end does blind transfer without going
on hold
Workaround: None.
8921: Centralized conference fails due to RTP port being slow to open in some
cases
Workaround: None.
9176: Memory leak in phone if it tries to upload log files into a non-existent
folder which is specified by LOG_FILE_DIRECTORY
Workaround: Specify a valid folder destination in LOG_FILE_DIRECTORY.
9292: IP 4000 reboots upon downloading a wave file with a path containing ‘\’
instead of ‘/’
Workaround: Wave file paths must be specified using ‘/’ e.g. “wavs/ring1.wav”.
9709: RTCP not sent or received when calls are on hold
Workaround: None.
11588: The local contact directory feature cannot be disabled
Workaround: None.
12155: SoundPoint IP 300 and 301 phones have no “Exit” softkey during the
ACD login process
Workaround: Exit the display by pressing the Menu key or lifting and replacing the handset.
12455: On SoundPoint IP 601 phone, per-contact directory settings such as
auto-divert do not work for calls arriving on lines 7 to 12
Workaround: None.
12492: SoundPoint IP 601 phone with Expansion Module(s) attached may fail
to load the selected language after rebooting
Workaround: Switch to English (Internal) and then back to the desired language after the reboot.
Copyright © 2006 Polycom, Inc. Page 25
Release Notes - SIP Application Notes
12616: Phone crashes after receiving high call rate (4 unanswered calls every 18 seconds)
Workaround: Reduce the incoming call rate.
12647: Feature keys cannot be reconfigured to perform other functions Workaround: None.
12722: Stuttered dial tone does not work if first line is shared
Workaround: Configure the first line on the phone as a private line
12952: There is no way to reset the user password back to the factory default password
Workaround: None.
13076: Phone can pause at the “Welcome” screen for more than 5 minutes after being rebooted
Workaround: Ensure that the boot server can handle the load of multiple phones rebooting.
13230: No audio on calls resumed from hold in some multiple call scenarios Workaround: None.
13412: Cannot edit the contact directory on the phone if the phone’s directory file saved on the boot server has been corrupted
Workaround: Correct the directory file on the boot server and reboot the phone.
13579: SDP parser applies wrong logic
Workaround: Change the order of lines in the SDP.
13786: HTTP Digest Authentication does not work on IIS
Workaround: Use a different form of authentication, a different protocol or a different server
14275: The call.callWaiting.prompt parameter does not have any effect Workaround: None. This functionality changed in SIP 1.5.
14400: Phone can take up to 30 minutes to boot when there are TCP timeouts Workaround: Ensure that the configured boot server is running correctly or do not use a boot server.
14466: 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.
14467: If a URL in <Ethernet Address>.cfg specifies a protocol and user name but no password, the password in flash is not used
Workaround: Specify the password in the configuration file
14624: Boot servers running explicit FTPS are not supported
Workaround: Use implicit FTPS or HTTPS.
14844: A failed download of a pre-existing file causes that file to be deleted
Workaround: None.
14937: Pattern generator for tones does not work well for the case of a single repeating chord
Page 26 Copyright © 2006 Polycom, Inc.
Release Notes - SIP Application Reference Documents
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.
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
Workaround: Include the password in the server address URL.
16041: After a reboot, a phone with a shared line is occasionally unable to
seize the line
Workaround: Reboot the phone again.
17102: IP430 locks up when performing a reboot on detection of a suspended
task.
Workaround: Manually reboot the phone.
4. Reference Documents
Administrator Guide – SoundPoint IP SIP – Version 1.6
Copyright © 2006 Polycom, Inc. Page 27
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