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
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
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
• 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)
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
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
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
• 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.
• 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.
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.
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)
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
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.
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
•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.
• 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
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