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KIRK Wireless Server 6000
Release Notes Q4 2014
Firmware Version PCS14C_
Release Notes
Q4 2014
KIRK Wireless Server 6000, Release Notes
Table of Contents
Introduction ...................................................... 5
Changes ............................................................................. 8
Release Notes
Q4 2014
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KIRK Wireless Server 6000, Release Notes
Revision History
Date
2008-06-16
2008-09-24
2008-11-04
2008-12-09
2009-03-05
2009-06-10
2009-08-04
2009-10-12
2009-10-19
2009-12-10
2010-01-27
2010-03-31
2010-06-21
2010-07-13
2010-10-11
2010-12-15
2011-03-08
2011-05-29
2011-06-14
2011-09-14
2011-12-12
2012-03-19
2012-06-12
2012-09-14
2012-12-10
2013-03-11
2013-06-11
2013-09-24
2013-12-16
2014-03-24
2014-06-24
2014-09-22
Description
First draft
Release notes for PCS02A_
Release notes for PCS02B_
Release notes for PCS03__
Release notes for PCS03A_
Release notes for PCS03B_
Release notes for PCS04__
Release notes for PCS04A_
Release notes for PCS04B_
Release notes for PCS05__
Release notes for PCS05A_
Release notes for PCS05B_
Release notes for PCS05C_
Release notes for PCS05D_
Release notes for PCS06A_
Release notes for PCS07__
Release notes for PCS08__
Release notes for PCS08B_
Release notes for PCS09__
Release notes for PCS10__
Release notes for PCS12__
Release notes for PCS12BC
Release notes for PCS12C_
Release notes for PCS12D_
Release notes for PCS13__
Release notes for PCS13B_
Release notes for PCS13E_
Release notes for PCS13F_
Release notes for PCS14__
Release notes for PCS14A_
Release notes for PCS14B_
Release notes for PCS14C_
Release Notes
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KIRK Wireless Server 6000, Release Notes
Release Notes
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KIRK Wireless Server 6000, Release Notes
Introduction
Release
These release notes apply to the released versions of firmware for the KIRK Wireless Server 6000
(hereinafter referred to as KWS6000 or KWS). This version specifically applies to version
PCS14C_ of the KWS6000 firmware. The release replaces the PCS14B_ release as the latest generally available (GA) release.
Important Notes
Some features require specific versions of the firmware loaded into the base stations or media resources.
Feature License and Platform Limitations
The following table summarizes features that require a particular hardware platform and/or a license key for activation.
Feature Comment
License required.
Part number 14075260
Software License for Backup KWS6000
Redundancy with ARI Swap License (from software version PCS06A_ or newer)
Software License for Master KWS6000
Redundancy (from software version PCS06A_ or newer)
KIRK Software Security Package (HTTPS, TLS,
SRTP)
License required.
Part number 14075250
License required.
Part number 14075280
KIRK Microsoft Lync Interop incl. KIRK Software
Security Package (HTTPS, TLS,
SRTP)
KIRK Automatic Alarm Call License
License required.
Part number 14075270
License required.
Part number 14075450
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KIRK Wireless Server 6000, Release Notes
System Requirements
Hardware
Hardware Platform
KWS6000 HW PCS 3C or newer
Media Resource 6000 HW PCS 3C or newer
Description
KWS6000 Server
Media Resource 6000
Software
The KWS communicates with media resources and base stations using a version controlled communication protocol.
When updating a KWS with new firmware, this might introduce a new version of the communication protocol towards either media resource or base station.
To minimize downtime when updating a KWS, media resources and base stations, it is recommended that you use the following approach.
Update all infrastructure units: base stations, media resources, and KWS to new firmware before rebooting any of these. The new firmware and thus the new protocol will not be active until the unit has been rebooted. When the firmware update of all units is successful, reboot the system in the following order:
Base stations first then media resources, and finally the KWS. The reason for this recommendation is that the base stations and the media resources can be rebooted from the KWS, which is much easier than logging into each unit manually. If the KWS is updated first, it might no longer be possible to control the base stations from the KWS.
Often new firmware of, for example, the KWS allows for
– but does not require - an update of media resource and base station firmware. The following table lists the firmware revisions of the
KWS that introduce new protocol versions and thus require an update of base stations/media resources.
KWS
Firmware
Media resource protocol
Base station protocol
Media resource
Firmware
Base station
Firmware
PCS13E_ 12 7 PCS13B_ PCS13E_
PCS13B_
PCS13__
PCS12D_
PCS12C_
PCS12BC
PCS08__
PCS06A_
9
8
6
12
11
11
10
5
5
6
6
5
4
3
PCS13B_
PCS12D_
PCS12D_
PCS12C_
PCS12BC
PCS08__
PCS06A_
PCS13A_
PCS13A_
PCS12B_
PCS12B_
PCS12B_
PCS08__
PCS06A_
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KIRK Wireless Server 6000, Release Notes
Distribution Files
Download the latest software for your KIRK Wireless Server 6000 at the Spectralink Support
Portal
. Sign up for Spectralink’s technical newsletter
Tech Point to get updated on new software releases and technical information.
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KIRK Wireless Server 6000, Release Notes
Changes
Version PCS14C_ – Q4, 2014
Added or Changed Features
Improved monitoring capabilities on radio interface.
The base station will monitor active radio connections and report frame errors in voice/data packets back to the server. This is utilized in the server to display the amount of frame errors on the individual base stations. The statistics are available in Statistics|Base Station, as well as in the exported logs. These statistics should be read with care, and some amount of frame errors should be expected. The statistics can be used to get a high level overview of the health of the installation/deployment HF-wise. If a base station has a high amount of frame errors compared to the other base stations, this can be an indication of e.g. a challenging
HF-environment or potentially a base station located at the edge of the coverage area with many handsets attempting connections outside the range of normal operation for the base station.
Improved monitoring capabilities on network interface.
The media handling will monitor active RTP sessions and report problems and packet loss to the server. This is utilized in the server to display the amount of RTP loss on the individual base stations and media resources. The statistics are available in Statistics|Base Station and
Statistics|Media Resource, as well as in the exported logs. The statistics can be used to get a high level overview of the health of the installation IP network-wise.
In addition to RTP loss statistics available for monitoring, a notice message will be logged if the number of RTP problems is greater than 2% and the duration of the connection is more than 5 seconds.
In addition to the above a lost packets counter is added to the current RTP statistics log message.
The statistics module has been augmented with statistics for devices. Every time a voice call is released Spectralink handsets supporting statistics, will send statistics to the server, which stores the information. The following statistics are available in the web GUI and in an exported log.
○ User
The username currently using the device. (If handset sharing is used and no user is currently signed-in to the device, the username may be empty).
○ Call time (hh:mm)
The total call time for the device (voice calls).
○ Abnormal releases
The percentage and the number of abnormal/total releases (voice calls).
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KIRK Wireless Server 6000, Release Notes
○ Bearer handover cancelled
The percentage and the number of cancelled/total bearer handovers.
A bearer handover is when the handset changes channel (frequency/timeslot) but stays on the same base station. This is typically performed if the handset detects disturbances on the current channel.
○ Connection handover cancelled
The percentage and the number of cancelled/total connection handovers.
A connection handover is when the handset changes base station. This is typically performed if the handset detects a better base station than the one it is currently on.
○ Call setup failed
The percentage and the number of failed outgoing call setups. An outgoing call setup can fail for different reasons. Some examples are: If the radio connection drops while establishing the call or if the user presses on-hook/off-hook quickly.
○ Text setup failed
The percentage and the number of failed outgoing text call setups.
○ Out of coverage
The number of times and the duration in hh:mm the handset has been out of coverage.
○ Frame errors
The percentage of voice/data packets with frame errors detected while in an active radio connection.
In addition to making the device statistics available for monitoring, an information message will be logged when a call is terminated if the frame error rate is above 2 %. E.g.
‘‘DECT errors reported by device, Username:1292
FrameErrors:4%(4235/91952)’’
When putting a call on hold the server needs a response from the other end (being put on hold). If this response is late and a second call is being initiated before the response is received the handset would enter an undefined state and the following message would be logged:
LogTriggerDump("SipCall [id:####] (UserName:####)
(SipEndpointRecvNGNewInd) unexpected state endpoint:#### call:####".
The server now handles this situation gracefully, the handset is returned to hold state and the message is no longer logged.
The abnormal releases statistics available at Statistics|Abnormal releases has been improved. Previously a user in an entry in this list was identified by a PPID, which is an internal identifier for users. Now the actual username is presented directly in the list for easier reference. Furthermore the IP DECT Server has been seen to restart, if a HTTP connection breaks while the abnormal releases list is being retrieved, this is no longer the case.
For solicited MWI accept NOTIFY before the OK response is received for the SUBSCRIBE.
This fixes an issue reported in DECTESC-541 where the MWI NOTIFY takes a different route
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KIRK Wireless Server 6000, Release Notes than the OK response for the SUBSCRIBE and thereby arrives before the OK response. This breaks the exchange of MWI messages and no MWI is displayed on the handsets.
The server now supports receiving item number from handsets on location registration. It is used to determine the handset name, when users and devices are displayed. This makes it possible to identify OEM branded handsets with the OEM naming.
In the WEB UI the possibilities for sorting columns with empty/non-numeric/percentage values has been improved. As an example the Administration|Base Station part of the WEB
UI now supports sorting base stations by synchronisation source (Prim/Sec) and by synchronisation lost counter. Furthermore the synchronisation lost column has been made easier to read by using a dash as separator between absolute numbers and percentages.
Make sure to terminate an incoming call correctly by sending a "480 Temporarily
Unavailable" response if a DECT error occurs while a media resource is being allocated.
RTCP handling has been improved.
RTCP lost reporting now supports negative loss, which the RFC allows. Furthermore RTCP lost calculations now only cover current SSRC, and lost statistics and RTCP reporting is only done for a SSRC when in sync.
SIP endpoint state/event errors were previously logged with log-level critical. This was misleading as a state/event error is unexpected behaviour however not critical. Starting with this firmware they are logged with log-level error.
Fix a confusing notice message when a non-INVITE transaction fails. If the remote User-
Agent/Server was unknown the message "Error received from unknown" was logged. Now the remote IP address is added to the message and "unknown" is not added to the message if the remote User-Agent/Server is unknown.
Previously state event errors like the following could be logged as a result of a scenario where the user presses a digit while a media resource is being allocated or while toggling between calls:
"SipEndpoint (UserName:XXXXX) (SipEndpointRecvCCInfo) unexpected signal/SIP in state ResourceAlloc/TwoCalls_OutgoingHold"
This is not an error, and the logging has been removed. This resolves DECTESC-546.
If a call is terminated while the media is being completed, the server would previously log the following error message:
"MR_COMPLETE_cfm received for unexpected call UserName:####.
This is not erroneous behaviour, thus the logging has been removed.
The server will no longer send SUBSCRIBE for MWI before SIP registration is successful. If a user was updated via the GUI or via provisioning the server could send a SUBSCRIBE for
MWI before a successful SIP registration. This is currently not causing problems, but it is an undesired behaviour.
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Removed Features
None
Corrections
Using software version 14B_ in a setup with handset sharing, has been seen to cause an error where no more status messages e.g. voicemail, call forwarding, etc. are shown on the display of the handsets. When this error occurs the following message can be found in the log:
"TRIGGER MESSAGE Hashtable duplicate insert:"
This behaviour was reported in DECTESC-547, and has been corrected.
If the server is configured to only support G.711 A law media codec
(Configuration|SIP|Codec priority setting), the server/media resource has been seen to become unresponsive, if it receives a G.711 u-law RTP packet even if G.711 A law has been negotiated. This was reported in DECTESC-549, and has been corrected.
Configuration File Parameter Changes
None
Version PCS14B_ – Q3, 2014
Added or Changed Features
Syslog settings are now controlled and distributed from the server. I.e. the settings on the server will override/control the settings on connected base stations and media resources.
Previously if a syslog server was used, every single unit of the infra-structure needed to be configured for syslog individually, now everything can be configured in one place (at the server).
This means that the following settings will be set on the base stations and media resources controlled by the server.
- log.syslog.host
- log.syslog.port
- log.syslog.facility
- log.syslog.level
The system time and date as well as the time zone is now controlled and distributed from the server. I.e. the settings on the server will override/control the settings on connected base stations and media resources.
When parsing SIP parameters, the server is now more forgiving with lower/upper case of some SIP parameters. This addresses DECTESC-534 where a "replaces" require was not
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KIRK Wireless Server 6000, Release Notes understood from a specific Lync endpoint. This resulted in problems making a transfer from
Zeacom (Lync SW based attendant console) to DECT handsets.
A second incoming call (a call waiting call) is no longer automatically accepted if the other end of the original call terminates the original call. If A and B are in a call and C calls A, A will receive a call waiting indication. Previously, in the case of B hanging up, this would automatically result in A accepting the call. This is not necessarily what A wants, thus this has been changed. Now A actively needs to accept the new call.
In an outgoing call, if a other party display name is received in a 1XX response to the outgoing INVITE, the other party display name is displayed in the handset. As an example this means that if A calls B, the display on A will indicate the display name of B already when phone B starts ringing. Previously the display name would not be updated before the call was answered by B.
The IP-DECT server now supports parsing a SIP Contact header with a comma in the URI.
This addresses an issue reported in DECTESC-535 regarding call transfer in a Shoretel PBX environment. The scenario causing the issue is as follows:
An incoming call from a mobile phone hits a reception workgroup and makes a number of
DECT handsets start alerting. The call remains unanswered and after a timeout, controlled by the Shoretel PBX, the call is automatically forwarded to a hunt group. A DECT handset, which is part of that hunt group, answers the call and attempts a consulted transfer to another phone. The above mentioned scenario failed before and is successful with the latest firmware.
The name and version string of the connected SIP Server/Registrar is now logged when available. As an example when an OK is Received for a SIP REGISTER the Server
Name/User Agent string received in the OK is logged in an INFO level log message. See the following for example log messages:
‘‘SIP user 2602 registered at 172.29.198.3 OpenSER (1.3.4-tls
(i386/linux))’’
‘‘SIP REGISTER response 404 Not Found from:2602 to:2602 Error received from Cisco-CUCM10.0’’
Some log messages indicating Signal link up/down were causing confusion because some customers interpreted these as if the IP link was up/down. It is now written as DECT signalling link up/down, to avoid confusion. Reported in DECTESC-530.
In a redundancy setup the slave did not log the reason for TCP connect failure with the master. This made debugging of failure scenarios difficult. Now the reason is logged.
A state/event error is no longer logged, if an outgoing call is put on hold while ICE is completed. This is not an error situation and should not be logged as such.
Dialog event package handling has been improved to provide more correct signalling towards e.g. Cisco Unified Call Manager.
Various improvements to TCP/ICE handling. This includes more elaborate logging in failure scenarios to facilitate debugging and better handling of problematic scenarios.
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KIRK Wireless Server 6000, Release Notes
Logging of media handling has been augmented, this has no impact on normal usage however it increases the possibilities for debugging media-stream related problems.
Removed Features
None
Corrections
If a call transfer to an entry in the phonebook was attempted on a 74-series handset, it was not successful. The signalling is now changed to allow for such a call transfer to succeed.
This addresses an issue reported in DECTESC-533.
Media handling has been improved. Make sure that resources and sessions are reset when allocated. This fixes an issue introduced in PCS14A_ which made handover and internal
SRTP not work reliable. If internal SRTP was enabled some handover scenarios would result in noise in the media.
The handling of handover signalling has been improved. In very rare scenarios, with handover occurring early in the call establishment, the handover could fail potentially resulting in loss of audio. This is now handled more gracefully.
The process of subscribing handsets without requiring a specific IPEI has been improved.
Previously if a SIP user was created without assigning a specific IPEI (device) and a handset attempted a subscription, the IPEI of the handset would be saved in the database even in the case of an unsuccessful subscription (e.g. if a wrong AC was entered). This resulted in the fact that any subsequent subscription attempts would require the same handset to be successful (otherwise the IPEI would not match). Now the IPEI is not stored in the case of an unsuccessful subscription.
If the IP DECT Server receives a Lync conference Notify from a Lync server, it now responds with a 481 (Call/Transaction does not exist) instead of 500 (Internal server error) if no Lync conference is present. This is more correct.
Configuration File Parameter Changes
None.
Version PCS14A_ – Q2, 2014
Added or Changed Features
IMPORTANT the default password for the WEB GUI has been changed from " ip6000" to
"admin". This change has been made in order to use the same default password for all current IP DECT infrastructure components. This change has no impact if the default password has been changed or if the Configuration|Security settings have been saved, in which case the default password would have been saved to the configuration file.
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KIRK Wireless Server 6000, Release Notes
The tables used to display information on the WEB GUI have been improved in order to better handle many users and base stations. The improvements include a search function, allowing searching all table data for a specific string, and a sort function, allowing sorting the table data with respect to any column. Finally, a pagination function, allowing breaking down the table data into suitable page sizes is now available.
Optimize the internal signaling in the IP DECT server to better handle if the SDP from the other end is received late in the call setup process for incoming calls. Typically SDP from the other end is included in the incoming INVITE. However in some PBXs/scenarios no SDP is received before the final ACK. Previously a delay in the audio setup could be experienced if the ACK was received very late. DECTESC-524 describes an issue with audio transmission starting late, after answering an incoming call on a DECT handset.
The IP DECT server WEB GUI now supports displaying the HW version, SW version,
Production ID and Production date for Spectralink handsets. The same information is available in the users.xml part of a log export generated by Status|Logs|Export. The information is sent by newer handsets at subscription & location registration.
The IP DECT server will, starting with this release, accept a new incoming call while an incoming call release is pending. Before this change the IP-DECT Server would reject the new incoming call with a busy response. Now, the new call is accepted and sent to the handset, as soon as the pending release is completed. The IP-DECT Server is required to accept this signaling in order to handle Cisco Unified CM call pickup and some kinds of hunt groups. For this to work perfectly a handset firmware update is required as well. If the handset firmware is not updated the new incoming call may be reported as an abnormal call release instead of a busy.
Release LCE (lower layer) instance correctly when an incoming call is released. The LCE instance was not released correctly when an incoming call was released and if a new call was received within a few seconds after the release, it ended as an abnormal release. This addresses DECTESC-522 where a hunt group initiates new calls to a handset shortly after the release of the previous call. In the specific use case reported, one of the users in a ring group is busy, then the ring group is being called and while the other handsets in the ring group are ringing, the busy member of the ring group terminates the call and gets
(immediately) a new incoming call, as it is part of the same ring group. Previously the IP-
DECT Server would respond "Temporarily not available"; now the IP-DECT server can handle the second incoming call.
Handle multiple 180 Ringing responses. In scenarios where an outgoing call is forked or forwarded multiple 180 Ringing responses may be received from different endpoints. The first ringing response was sent to the handset but any additional ringing responses were dropped.
This caused the handset display not to be updated correctly. Now all ringing responses are sent to the handset and the display will be updated correctly.
Change the WEB_GUI date display format from DD-MM-YYYY to YYYY-MM-DD. The new format is more standardized and is less ambiguous especially for users in the US.
In some call tear-down scenarios the IP DECT server previously potentially would attempt to terminate a media session after the termination of the DECT Call using the media session.
This would result in logging a "Session Free: lid (##) not in use" message with level critical.
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KIRK Wireless Server 6000, Release Notes
Now the IP DECT server will skip releasing the media session when it has been released by the DECT call.
Call state handling has been changed. When a SIP CANCEL request is received early in a call attempt to release the link layer with a LinkRelease instead of attempting to release the call control layer with a CCRelease.
Make sure that STUN information sent to MR is correct if UDP or TCP parameters are missing from the Lync server. Correct handling of missing UDP or TCP address for TURN.
With this change the IP DECT Server is more robust towards missing UDP or TCP parameters in STUN/TURN parameters provisioned by a LYNC server.
The IP DECT server will only accept Microsoft Lync conference invitations if they contain audio. Previously the IP-DECT Server did not check for audio in a conference invitation before it was accepted and this caused unintended behaviour.
Show phase/offset in neighbours list. This value represents the offset between a base and the other bases (neighbours), it can see over the air. Handsets (and base stations) can compensate for measured offsets of +/- 10, thus a large offset is not necessarily a problem, however the neighbours with strongest signal strength (RSSI) should have an offset within
+/- 10.
On incoming text calls ignore 0-9,*,# to align with DECT Server 2500/8000. Before this change pressing these keys before pressing ok to accept the incoming text call would result in the release of the text call with cause 84. This is based on feedback from Connexall.
Pass R,X,O,H through before a MSF text call is accepted in order to signal what button the user pressed to release the text call. On request from Connexall. This can be used by an application to determine why a text call was released if not by pressing ok or reject (i.e. hook, etc.)
Adapt to new time zone for Moscow. Moscow is no longer using Daylight Savings Time, and is fixed at GMT+4; this is now reflected in the time-zone string for Moscow.
Minor changes to WEB_GUI for administering base stations. The Administration|Base station table has undergone minor layout changes and no longer displays which synchronization source is currently used for synchronization.
Respect CLI restrictions sent from Cisco Unified CM in Remote-Party-ID header. The
Remote-Party-ID header has a privacy part, which can be used to control whether or not
Display Name and Number should be displayed by the endpoint. If either are marked as restricted the IP-DECT server will not send them to the handset.
The IP DECT Server now supports handling SIP bodies starting with \n. Previously this was not supported and Cisco Unified CM has been seen to send SIP bodies starting with \n in connection with remote call control.
Incoming early media handling has been improved. The IP DECT server now handles a=inactive in SDP in a 1XX response.
Added a new setting to the GUI: "TCP ephemeral port in contact address". Enable this to add the TCP ephemeral port (the local TCP port of the outgoing connection) to the contact header used in outgoing SIP messages. Otherwise the local listening port is used.
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KIRK Wireless Server 6000, Release Notes
Add the correct port number to the SIP contact header when configured with a local port different from 5060 or configured with endpoint separate ports enabled. Without this correction no port number was added and the default 5060 was assumed.
Delete GRUU when registration data is cleaned up. When connected to a SIP server that supports Globally Routable UA URI (GRUU) a GRUU is retrieved as part of the registration process. The GRUU should be deleted when the registration data is reset, because a new one is received on the next successful registration.
Starting with this release the IP DECT Server will support a maximum of 32 allowed peers in a forked call. Previously a maximum of 8 were supported. One use case affected by this change is an outgoing call from a DECT phone to a Lync user who has a ring group (either
Team group or Delegate) that rings multiple Lync users. Previously if the group had more than 8 members and no-one answered, the call would be diverted to voicemail after some period of time and the DECT phone making the call would display "Media Failed
”. Now - the call is transferred successfully to voicemail as long as the number of members in the group is
32 or less. This issue was reported in DECTESC-477.
Handle XML escape characters in description when generating rfps.xml.
Do not log a notice message when "481 Call Leg/Transaction Does Not Exist" is received for
BYE. This is not uncommon in transfer scenarios and is not a problem.
Degrade log message from warning to debug when the HTTP connection is lost while displaying the log. This is not uncommon and not a problem for the system.
Removed Features
None
Corrections
Use the correct component index when creating TCP srflx candidates and correct cleanup of srflx candidates. This resolves an issue reported in DECTESC-515 which describes a situation where it is not possible to make calls from a DECT handset to a Microsoft Lync client.
Save SRTP authentication parameter so we do not trigger a new SRTP creation on each configure. Furthermore do not create new SRTP if no parameters have changed. Before this change sequence numbers on SRTCP could get out of sync with what a Mediation server would expect, which could lead to the disconnection of calls. This resolves an issue reported in DECTESC-528. In a Microsoft Lync setup if an external user is on-hold from a DECT handset, the external party might be disconnected after 30 seconds.
Fix a bug with SIP dialog event package which is used for Busy Lamp Function (BLF) and by
Cisco Unified CM. When an outgoing call was initiated the NOTIFY dialog event was sent before the dialog was initialized and data was invalid.
When sending peer reflexive ICE candidates, add correct relative address and select correct candidate for RTCP. This resolves an issue reported in DECTESC-527 concerning Skype calls in a setup with Lync 2013 and IP-DECT server. An external Skype user calls over
Release Notes
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KIRK Wireless Server 6000, Release Notes federation service and the call can be answered with a Lync Client or a handset on the IP
DECT server. If the call is answered with the DECT device, the call would terminate after a few seconds.
Reset in-memory SIP authentication data when user parameters related to authentication is changed. The authentication mechanism was not reset correctly when user data with impact on the authentication was modified. This potentially had the effect, that administrator changes did not take effect until the system was restarted, or the user was disabled and enabled again.
Previously the IP DECT Server could in some cases attempt to access TCP connection data on TCP connections which were no longer valid. This has been corrected.
Configuration File Parameter Changes
File Action Parameter
config.xml Added sip.tcp_contact_ephemeral_port
Description
Enable this to add the
TCP ephemeral port (the local TCP port of the outgoing connection) to the contact header used in outgoing SIP messages. Otherwise the local listening port is used.
Values: true/false
Default: false.
Version PCS14__ – Q1, 2014
Added or Changed Features
The DECT Server is now able to handle call forking to a mix of ICE and non-ICE enabled endpoints. Previously, a scenario with forking to a mix of ICE and non-ICE enabled endpoints could result in the disabling of ICE for the complete call. This resulted in issues with calls from DECT handsets to Lync clients. Sometimes, the call would not be established and an
ERROR 500 was displayed in the DECT handset display, when the call was answered. This issue has been seen in the field in a Microsoft Lync configuration, and was reported in
DECTESC-507.
The server no longer re-transmits SIP requests when TCP or TLS is used as transport protocol. When a reliable transport is used, SIP requests are not allowed to be retransmitted on timeout errors. Microsoft Lync was not able to handle these incorrect re-transmissions and spurious un-explainable errors occurred as a side-effect.
Handover is now aborted if the media resource fails to start the handover. This aborts the handover faster and more gracefully, and allows the server to log an error if it happens.
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KIRK Wireless Server 6000, Release Notes
Simplified the system backup/restore functionality. Earlier, parts of the complete backup could be restored individually. However, this could cause inconsistencies in the restored data due to interdependencies between different data such as users and DECT subscription.
The protocol for communication between the master and the slave server in a redundancy setup has been made more robust. Furthermore, there was a problem when more than 1000 users were replicated. This issue has been fixed.
If the handset and server call hold state get out of sync, it is handled gracefully, and does not result in the logging of an error. Without this correction, the handset could get stuck in a call hold state.
In the case where a media resource or a base station loses the connection to a server, the device will initiate a new connection to the server. The server will now accept a new connection from a device, before the server has detected a connection failure from the same device. This eliminates the “Already connected on socket” error, and reduces the time it takes the new connection to be ready for use.
Since ICE was introduced in the KGAP, an abnormal call release in an incoming call would be logged with “base: Unknown”, until the first handover was performed. Now, the time window where an abnormal release can report an unknown base station has been reduced drastically.
Furthermore, information about which base station the current call is on is now improved.
Previously, the part of the server that logs an abnormal call release did not know which base station the current call was on for any outgoing call before digits were received. For example, any abnormal call release in an outgoing call that was initiated by pressing off-hook (overlap dialing), resulted in an abnormal call release log entry with base: Unknown.
Do not check for blacklisting for the current registrar when connected to Lync or “Send all messages to current registrar
” is enabled. Some SIP servers do not support this and require a re-registration to switch to another proxy.
The system ARI is now displayed in the List Users part of the web-GUI. This way, the ARI is right at hand when subscribing handsets while monitoring progress on the List Users GUI page.
When exporting users to a CSV or XML file using Users|Import/export, users are sorted by username instead of by IPEI.
The procedure for delivering standby texts to handsets has been refactored and improved. It was previously allowed to update the standby text on the handset even in the case where an application was connected to the handset. However, this would tear down the connection between the application and the handset. As a result, changing the standby text from the GUI while the handset was in an active MSF call (e.g. corporate phonebook) meant that the server would end up in a state where it was not possible to update the standby text on the specific handset.
An IPEI is required before a SIP user registration is performed.
The free text search in the List Users part of the web-GUI now supports special characters.
This, for example, allows searching for a user with special characters in the display name.
Release Notes
Q4 2014
18
KIRK Wireless Server 6000, Release Notes
Take care of duplicate RPNs when performing synchronization loop check. Previously duplicate RPNs would confuse the loop check algorithm which would potentially lead to misleading results such as the reporting of false positives.
When configuring a base station as freerunning, set primary and secondary sync to its own
RPN and not 0. This is less misleading.
Removed Features
The Auto-create users functionality controlled by the Configuration|Wireless
Server|Autocreate users setting has become obsolete and has been removed. If a handset with an unknown IPEI is to be allowed to subscribe to the system, the recommended approach is to create a SIP user without specifying an IPEI. When subscribing a handset with an unknown IPEI, it is automatically created/associated with the first available SIP user without a specified IPEI.
Corrections
If the "force https" setting was enabled to ensure that exclusively encrypted access was available to the user interface, an unwanted side effect would occur. The export logs functionality would result in the export of empty log files. Specifically all HTML and XML files would be empty. This issue was reported in DECTESC-500. Additionally, it was not possible to perform a central firmware upgrade of media resources and base stations from the server if "force https" was enabled on the server.
Fixed a problem where ACK send to a hostname could not be TLS validated because the hostname was reset during DNS resolution. This error was seen in some Microsoft Lync installations, when the Lync server failed to answer quickly enough.
Avoid a buffer overflow and log a notice message if more than 20 ICE candidates are received in SDP.
If the server received an excessive amount of ICE candidates in an incoming SDP message
(more than 20) the server could restart. Now, if more candidates are received than the server can handle, the remaining ICE candidates are skipped and a notice message is logged to the message log. This issue was seen in the field in a Microsoft Lync setup and was reported in
DECTESC-496.
In a setup with redundancy, enabled changes to the base station synchronization settings were in some cases not replicated correctly between the master and the slave server. As a result changes to the synchronization chain would not take effect until the base stations were rebooted.
Configuration File Parameter Changes
File Action Parameter
config.xml Removed dect.auto_create_users
Description
If enabled, a user is automatically added to the
Server when a DECT handset tries to subscribe
Release Notes
Q4 2014
19
KIRK Wireless Server 6000, Release Notes
File Action Parameter Description
to a DECT system.
Values: true/false
Default: false.
Version PCS13F_ – Q4, 2013
Added or Changed Features
The KIRK/Spectralink handsets have been renamed according to the following table.
Previous model name
KIRK 4020
KIRK 4040
KIRK 4080
KIRK 5020
KIRK 5040
KIRK 6020
KIRK 6040
KIRK 7010
KIRK 7020
KIRK 7040
KIRK Butterfly
KIRK Site Survey
New model name
Spectralink 7420
Spectralink 7440
Spectralink 7480
Spectralink 7520
Spectralink 7540
Spectralink 7620
Spectralink 7640
Spectralink 7710
Spectralink 7720
Spectralink 7740
Spectralink Butterfly
Spectralink 7000 Site Survey
The integration with Microsoft Lync has been improved. Users on the KWS now support the
Microsoft Lync framework for being invited to a Lync conference. When a user is invited to a conference, the following three steps are involved.
Step one: When a Lync client invites a user to a conference, a special conference invitation is received by the KWS. This is the message that goes from the “inviter” to the “invitee” giving the URI of the conference focus. The message is a special SIP INVITE request with information about the conference instead of normal Session Description Protocol content in the message body.
Release Notes
Q4 2014
20
KIRK Wireless Server 6000, Release Notes
Step two: The KWS will start a signaling session with the conference focus, which is accomplished with another INVITE wit h a special content type of “application/cccp+xml.”
CCCP stands for Centralized Conference Control Protocol, which is the protocol Lync uses for communication with conference server roles. Once this session is established, the Lync user is connected to the conference, but does not yet have any media sessions established.
(This is the point where you can see someone in the conference roster, but the phone icon,
IM icon, etc. next to their name are still grayed out.).
Step three: The last step is to connect to the conference media. For an audio conference, this means dialing in to the audio/video MCU. The MCU, or Multipoint Control Unit, is the
Lync component that mixes media for the conference and distributes it to the participants.
When connected, media flows directly to the MCU. The last step is accomplished with a normal INVITE with Session Description Protocol content used to negotiate media transmission between the Lync user and the A/V MCU. Once this signaling session is established, and media begins flowing between the Lync user and the MCU, the conference join is complete.
The implementation of TLS/SSL authentication of clients connecting to the KWS has been updated. The previous implementation did not handle clients that start with a SSL2/3 handshake well. This problem was identified in the field caused by Perl scripts connecting to the KWS for supervision purposes.
If the media resource is local (located on the KWS), the link to the loopback address in MR administration has been removed.
The handling of Network Time Protocol (NTP) has been updated. The amount of time that can be adjusted is increased from 200 milliseconds to 1 second. Furthermore the minimum delay for the NTP response filter is increased from 10 milliseconds to 20 milliseconds. This gives a smoother operation of the NTP handling and should reduce the number of NTP notice messages in the message log, especially in scenarios with a jittery IP connection to the NTP server.
The messaging handling in the KWS has been refactored, and range checks for message text and callback numbers have been improved. The KWS is now prepared for handling longer text messages and longer callback numbers. Furthermore the KWS now supports sending more information in a single DECT protocol message than before. The maximum length of callback numbers is now 64 characters (previously 24). The maximum length of
MSF messages is now 180 characters (previously 72). Whether to use old or new limits is controlled by the setting dect.allow_long_messages. If this setting is false (default) the old limits are enforced. WARNING this feature is not yet supported by the handsets, thus setting dect.allow_long_mesages to true is not recommended at this point.
During a transfer the KWS now sends a BYE from the transferee to the transferor after the call to the transfer target has been established. This shortens the window where the transferee is not able to handle REFERs etc. from the transfer target. The old behavior can be restored by setting sip.send_bye_with_refer_notify=false. This solves DECTESC-485 SIP
Group overflow transfer not working. The issue was identified in an interop test with Shoretel.
Release Notes
Q4 2014
21
KIRK Wireless Server 6000, Release Notes
The media resource sends the build number of the running firmware to the KWS in the startup message.
The KWS Web-GUI now shows the build number of the firmware on Media resources and
Base stations in the case of development/beta versions of the firmware.
The KWS no longer responds with 400 Bad Request when a SIP server is terminating a SIP subscription. Specifically, Microsoft Lync sometimes terminates the SIP subscription for presence, and the KWS should respond with 200 OK. The wrong response can have caused problems in some presence scenarios.
Make remote syslog work with dynamically changed IP address. Previously, if the device changed IP address during operation, it would stop sending remote syslog messages.
Log an error on the KWS, if an uplane cannot be requested or connected on a base station.
Earlier this was only logged locally on the base station.
Log info message when provisioning download is started in order to ease debugging.
Make the RFP number look the same in the log messages. When referring to RFP number use RfpNo not RFP.
Make Status|Logs use HTML-safe formatting for log messages. Some messages might contain data which break HTML formatting. This is now escaped correctly.
Removed Features
None
Corrections
On handsets running in legacy mode (old generation user interface), the call waiting indication is now turned off correctly, when the remote end cancels the call. When an incoming call waiting is pending, the display shows an indication and an audible indication is played. This indication was not correctly cleared when the remote end cancelled the incoming call.
Configuration File Parameter Changes
File Action Parameter
config.xml Added dect.allow_long_messages
Description
This setting controls usage of long MSF messages and long callback numbers. If this setting is false (default) the following limits are enforced.
- 24 char callback number
- 72 char msf messages.
If the setting is true the following limits are enforced:
- 64 char callback number
- 180 char msf messages.
Release Notes
Q4 2014
22
KIRK Wireless Server 6000, Release Notes
File Action
config.xml Added
Parameter
sip.send_bye_with_refer_notify
Description
WARNING use with caution.
If the handsets do not support long callback numbers and messages, enabling this feature might cause handsets to crash.
Values: true/false.
Default: false.
This setting controls KWS behavior during transfer. If the setting is true the KWS will send a BYE from the transferee to the transferor after the call to the transfer target has been established.
Values: true/false.
Default: true.
Version PCS13Eb
Added or Changed Features
None
Removed Features
None
Corrections
Increase buffers used to create XML for SERVICE requests send to Lync. This fixes a problem discovered by Microsoft during interop test. The KWS was unable to make federated calls.
When moving remaining TCP buffer also move null termination. This addresses DECTESC-
480 TCP problem with Avaya. The scenario leading to this issue was related to group call on an Avaya Communication Manager (ACM).
Delete UDP connections from connection table when they are deleted. This addresses
DECTESC-430 where SIP stops to work after some connection trouble between redundancy master and slave. Furthermore cleanup connection creation.
If a call is set up on an existing MAC connection, re-use the old uplane if present. Solves
DECTESC-471 and DECTESC-483, DECTESC-476, DECTESC-481 and DECTESC-487 which all report about problems with losing audio after a handover.
Avoid underflow when subtracting unsigned lengths, which would cause large buffer overruns in RTP queues. One scenario which has been seen to trigger this behavior is related to changing audio-codec mid-call. This issue was reported in DECTESC-488.
Release Notes
Q4 2014
23
KIRK Wireless Server 6000, Release Notes
Do not try to create IPv6 sockets if IPv6 is not loaded by the kernel.
Media/RTP handling has been updated. The media load scaling used to calculate the available number of free channels on a base station, did not handle the simultaneous allocation of many channels correctly.
Configuration File Parameter Changes
None
Version PCS13E_ – Q3, 2013
Added or Changed Features
Improved handling of SIP server errors and failover. To improve the user experience and make the SIP communication smoother, the KWS now keeps track of IP addresses that cannot be reached due to transport or timeout errors. This is accomplished by blacklisting IP addresses with transport or timeout errors for 30 seconds. No communication with a blacklisted IP address will be attempted if an alternative (failover) address can be determined.
Specifically, this addresses DECTESC-441 where the KWS was unable to register all users to the secondary SIP server because it spent too much time waiting for timeouts from the primary SIP server.
Support changing IP address during operation. If the IP address is changed during operation by DHCP, the KWS now handles this gracefully. In previous versions of the software, this caused internal inconsistencies which could lead to malfunction.
Bump base station protocol version to 7. The base station protocol version is bumped in order to support IPv6 with KWS6500.
Change SIP User-Agent string format to better fit RFC3261 and make Microsoft Lync client version filtering possible. When configured for standard SIP, the User-Agent string is now product/PCSrevision. When configured for Microsoft Lync the User-Agent string is now product/major.minor.update.revision.
For example:
○ Standard: KWS6000/PCS13E_43123.
○ Lync: KWS6000/13.5.0.43123.
This addresses DECTESC-468.
When parsing SIP Alert-Info header used for controlling internal/external ringing and autoanswer, the KWS now matches part of a string instead of the complete string.
The following is, for example, now allowed:
Alert-Info: <http://www.vertical.com>;info=alert-external
This was reported in DECTESC-467 which refers to the Wave PBX 4.0.0.2780 from Vertical.
Release Notes
Q4 2014
24
KIRK Wireless Server 6000, Release Notes
Add P-Preferred-Identity to OK response for UPDATE request when configured for Microsoft
Lync. This addresses DECTESC-466 where external calls through an AudioCodes gateway is terminated after 30 minutes due to a failed session refresh from the gateway.
Remove P-Preferred-Identity from OK response for PRACK request when not configured for
Lync.
Do not discard incoming SIP requests without a username in the request URI. This is correct behavior. Specifically, this makes the KWS answer correctly with a 501 Not Implemented error when a REGISTER request is received.
Change the TCP port used for communication between redundant servers to 56017. Port
58017, which previously was used for this, is in the default range for media resource external
RTP. The new port 56017 is in the same port range as the other TCP signaling channels.
More correct registration of the base station number (RfpNo) when an abnormal call release is logged. In some situations, the current base station is not known. In that situation, it was logged as RfpNo=0. This made RfpNo 0 take the blame for more abnormal releases than was correct. Now, the base station is logged as unknown when it is not known.
When no time zone has been configured the KWS will use UTC. Previously, the
Configuration|General|Timezone in the GUI would display UTC-
1 (Amsterdam,Barcelona,…) even though the timezone was unconfigured. This has been changed.
Ensure that the MAC address and UUID of the device is always handled and presented in lowercase.
During a handover with DECT encryption enabled the encryption parameters are exchanged earlier in order to make a smoother handover.
Prepare for new base station features and store more information about the base stations in the KWS base station database.
Make the protocols between the KWS and the media resource and the base station more robust. This is to reduce the probability that a problem in one of the units will cause problems in the other units.
Removed Features
None
Corrections
Correct handling of (S)RTP when an outgoing call is forked to a mix of endpoints using RTP and SRTP. The KWS switched correctly to SRTP but failed to switch back to RTP if the call was answered by an endpoint not using SRTP. This addresses DECTESC-444 where a call in a Microsoft Lync setup is forked to a PSTN gateway not using SRTP and some Lync clients using SRTP. If the call was answered by the gateway, noise was played by the gateway.
Fixed a problem where an incoming call never completes the STUN/TURN/ICE allocation.
The problem is solved by not using STUN/TURN when ICE is not used. Specifically, this
Release Notes
Q4 2014
25
KIRK Wireless Server 6000, Release Notes addresses DECTESC-461 where an incoming call from a PSTN gateway in a Microsoft Lync setup never makes the DECT handset start alerting.
Improve the quality of the transmitted RTP towards external endpoints when a CODEC card is mounted. A problem with some internal buffers made the media resource skip some frames causing bad sound quality. This addresses DECTESC-465.
Fixed a problem where the KWS released a waiting call towards the handset even though no call waiting was signaled to the handset. This resolves DECTESC-462 where a Lync team call scenario caused the handsets to generate an abnormal call release due to invalid signaling from the KWS.
Allow connections to the HTTP and XML-RPC server from clients using TLS 1.1 and newer.
Without this fix, recent versions of, for example, Google Chrome failed to connect via
HTTPS.
Correct check for synchronization loops when a base station is configured for auto sync.
Without this correction, base stations configured for auto sync. were not checked for sync. loops and undetected sync. loops could exist.
Auto sync. is still not intended for production use and should only be used during the deployment phase.
Removed a state/event error when MR_COMPLETE_cfm is received for a terminated call.
This could happen if a call was terminated very shortly after it was established.
Terminate call correctly if MR_COMPLETE_cfm returns bad status. In some situations, when the media negotiation failed to complete correctly, a BYE was not sent and the remote endpoint had a hanging call.
Remove memory leaks and static code analysis problems and make the software more correct and robust.
Configuration File Parameter Changes
None
Version PCS13B_ – Q2, 2013
Added or Changed Features
The KWS integration to Lync has been improved. The KWS now supports controlling the global call forward state of a Lync user. Handsets connected to Lync via the KWS are now able to manipulate the global call forward state of a Lync user, using configurable feature codes. The default feature codes are:
Feature code Decription
*21*<extension>#
*21*
Enables unconditional call forward to extension
Enables unconditional call forward to voicemail
Release Notes
Q4 2014
26
KIRK Wireless Server 6000, Release Notes
Feature code Decription
#21# Disables call forward
The feature codes are configurable under Configuration -> Wireless Server.
The global call forward state is reflected in the handset display by pre-pending the standby text with either [CFU] for unconditional call forward or [CFM] for call forward to voicemail.
The KWS will no longer require a NOTIFY event vnd-microsoft-roaming-self from Lync to be in a dialog. Lync will in some scenarios send a NOTIFY before the 200 OK that creates the dialog. This resolves DECTESC-440 where the KWS did not present the correct callingparty-number.
Only allow INVITE with replaces for the inactive/secondary call while SIP endpoint is in a two calls state. A strange Lync transfer scenario caused a crash because INVITE with replaces was received during a transfer.
The media handling in handover scenarios has been optimized.
During a handover the Media resource will stay on the original base station as long as media is received from it. Previously, the media resource could in some cases switch to the new base station too early, which could result in a short crackling noise during handover.
Furthermore, SRTP handling during handover has been updated. The Media resource could previously detect false replay attacks, during handover when SRTP was received from two base stations. This is no longer the case.
Previously, the KWS would log a state/event error in some scenarios where a handset ended a call which was put on hold. This is no longer the case.
The Polycom branding and naming in the product has been removed and replaced by
Spectralink. Theme handling for customizing naming and web-interface has been added.
IP address configuration GUI has been changed. IPv4 boot protocol selection (DHCPassigned or static IP address) is now done using a combo box instead of two radio buttons.
Additionally some more tooltips have been added to network settings.
The VLAN setting has been moved from IP settings to Ethernet settings.
LED handling has been changed.
Previously, the indication LED on the KWS flashed green, if a voice call was active. Starting with this release, the LED flashes green if a radio connection is active, even if no voice stream is established. This makes LED indication consistent for KWS and base stations.
If the device is a Media resource, the LED flashes green if a voice stream is active on the device.
The default Certificate Authority (CA) bundle with trusted CAs has been updated.
The rfp tag in the rfps.xml file now has a description property. This way the textual description will be available in the rfps.xml file.
The conversion from handset part number to textual description used in the Users | List
Users part of the WEB-GUI has been updated.
Release Notes
Q4 2014
27
KIRK Wireless Server 6000, Release Notes
The new KIRK 4080 (14122802) has been added, and several ATUS part numbers have been renamed.
Tooltip for capture PCAP custom filter now says to use PCAP filter syntax.
Removed Features
None
Corrections
Fixed problem with Lync presence publication. The KWS could sometimes stop publishing the presence state of a handset and consequently the Lync user could appear as always busy. The problem was caused by incorrect handling of incoming presence NOTIFY requests without state information.
The problem was reported in DECTESC-447. It was introduced in firmware PCS12C_ as an unwanted side effect of introducing support for SBA.
Fixed problem with Statistics Abnormal Releases list, when running redundancy. Previously, the reported total did not include abnormal releases from the slave. Now the total is correct, and the abnormal release list is sorted by timestamp (and not by master first, then slave).
This addresses DECT-222.
Mark central phonebook update as idle when a LDAP update fails. This corrects a problem where the phonebook stopped updating after a failed LDAP access and a reboot was required.
Previously, if a user with a username longer than 32 characters initiated a text call, it could result in a restart of the KGAP, for example when accessing the central phonebook. This has been corrected.
Fixed a rare problem with user database replication from master to slave in a redundant setup. If usernames or IPEIs are interchanged for two users while the slave is disconnected, the slave will not be able to store the changes and responds with an error duplicate username or IPEI. This is fixed by automatically deleting the slave user database and rebooting the slave.
Log a critical error if the master is unable to store data in the slave during replication at connect.
The problem was reported from the field.
Supports deleting clusters when replicating from master to slave.
Eliminated an issue seen with redundancy and XML-RPC. If, for example, an XML-RPC application sent an SMS with an unknown username, and the KWS was running a redundancy setup, it would result in a restart of the KGAP.
Removed memory leak in the central phonebook when empty strings are retrieved from
LDAP.
If a license was installed which allowed the KWS to handle more users than supported by the
KWS, the number of allowed users was displayed incorrectly. This has been corrected.
Release Notes
Q4 2014
28
KIRK Wireless Server 6000, Release Notes
Configuration File Parameter Changes
File Action Parameter
config.xml Added feature_codes.call_forward.voicemail.enable
Description
Specifies the feature code used for enabling call forward to voicemail (CFM) on
Lync.
Values: The feature code users must dial to enable call forward to voicemail.
Default: *21*
Version PCS13__ – Q1, 2013
Added or Changed Features
Support for 802.1p VLAN Class-of-Service tagging has been added. It is now possible to control the 802.1p Class-of-Service Priority Code Point (PCP) which refers to the IEEE
802.1p priority. It indicates the frame priority level. These values can be used to prioritize different classes of traffic (voice, video, data, etc.). The PCP can be set individually for signaling and media (RTP). For a discussion of recommended values refer to http://en.wikipedia.org/wiki/IEEE_802.1p.
The "Play on-hold tone" setting that controls whether the handset should play a tone when it is on hold was previously only supported if the handset was using legacy (old) user interface.
Handsets using context sensitive user interface did not play a tone regardless of the setting.
This behavior gave some problems on, for example, CUCM which does not send on-hold music, now the tone setting is supported also when using context sensitive user interface.
Context sensitive user interface is used on the following handset models/Firmware versions.
Handset model Firmware version with context sensitive user interface:
Kirk 30xx series
Kirk 40xx series
Kirk 50xx series
Not supported
Not supported
PCS12Ta and newer
Kirk 60xx series
Kirk 70xx series
Kirk Butterfly
Non Kirk handsets
PCS12Ta and newer
PCS12Ta and newer
All versions
Not supported
Release Notes
Q4 2014
29
KIRK Wireless Server 6000, Release Notes
Previously if the KWS received a 1xx response with SDP it would wait for RTP to be received from the other end before sending RTP to the other end. In scenarios where the KWS and the proxy/other end are separated by a NAT/Firewall the other end cannot send RTP through the NAT/Firewall before the KWS has sent RTP. Specifically the above behavior resulted in the fact that no early media was received from the other end. The issue was reported with hosted Broadsoft and Asterisk installations and was reported in DECTESC-437. Now the
KWS will send RTP (silence) to the other end as soon as a 1xx response with SDP is received, which will establish a connection through the NAT/Firewall allowing for early media to flow the other way. An exception of the above behavior is if ICE is involved in media handling (typically only the case in MS Lync installations).
Previously, the KWS was able to handle called party number and calling party numbers with a maximum of 36 characters. Starting with this release the KWS is able to handle called/calling party numbers with a maximum of 64 characters. Realistically the only scenarios where a Calling Party No of more than 36 characters is encountered will be if it is a
SIP URI: sip:[email protected] which might be the case in, for example, a MS Lync setup.
If called/calling party numbers with more than 36 characters are used, older handset firmware revisions might not be able to handle it. To ensure that the handset can handle more than 36 characters, use the following handset firmware:
Handset model Firmware version with support for more than 36 character Calling/called party numbers:
Kirk 30xx series Not supported.
Kirk 40xx series
Kirk 50xx series
Kirk 60xx series
Kirk 70xx series
Not supported.
PCS13__ and newer.
PCS13__ and newer.
PCS13__ and newer.
Kirk Butterfly PCS13__ and newer.
When the KWS receives an incoming call, the new 64 character limit for calling/called party id together with a potentially long display text might result in a signal which is too large to be sent to the handset.
If the KWS receives such an incoming call, the display text part of the signal might be truncated or potentially removed completely.
If this happens, an INFO message will be logged in the log, describing what was truncated.
In (semi-)attended transfer situations the KWS acts more correct in accordance with
RFC5589. This is required by some SIP servers and it eliminates an annoying log notice message when a transfer is performed between KWS handsets on some SIP servers.
Specifically, make the KWS acting as transferor send BYE to the transferee on reception of
NOTIFY with 200 OK. Earlier the BYE was send by the KWS acting as transferee together with the NOTIFY with 200 OK.
Release Notes
Q4 2014
30
KIRK Wireless Server 6000, Release Notes
The handling of the message log has been improved for better scalability and better performance. If an update of the message log is requested from the WEB-GUI (Status | logs) the display filter level is used in the log daemon instead of filtering messages in the HTTP server. This way a very high number of log messages of low level, for example, level INFO, do not need to be sent to the HTTP server if the display filter level is e.g. WARNING or
ERROR. Instead only the messages of the requested log level and higher will be sent to the
HTTP server. This gives a more responsive GUI experience when using the display filter level.
The KWS will now print single-line log messages on the console and in the txt log file, for improved readability.
When sending a SDP offer with the attribute sendrecv but receiving a SDP answer with the attribute inactive or sendonly, the KWS will now adhere to this and not send RTP. This is more correct and eliminates sending RTP that the far end will not handle anyway.
To maximize uptime and eliminate scenarios where it is necessary to restart he KWS, changing the setting for media TOS does no longer require a reboot before taking effect.
The KWS will no longer attempt to extract additional DNS records from a DNS SRV answer.
The feature was not working as intended and only one additional record was extracted even though more records might be present. With this fix the additional records are not used, and instead a new DNS request is issued and all records will be taken into account.
The size limit of internal signals within the KWS has been increased in order to be able to handle huge signals with ICE data consisting of (amongst other things) many IP addresses.
On the SIP transaction layer the Via and Contact headers are initialized with dummy values. The correct values are now added by the SIP transport layer.
The SIP routing logic has been improved/simplified.
IP address handling has been changed. To allow IP address changes during operation determine the local IP address on a per connection basis instead of for the device. This prepares for multiple simultaneous IP addresses on the device and allows a DHCP server to re-assign the device IP address without a reboot.
While changing the IP address during operation (i.e. without rebooting the KWS) is not recommended it can never-the-less happen in some scenarios. The KWS cannot control whether the DHCP-server decides to change the IP address it hands out (a rebooting DHCPserver might do this). Another example is a scenario with a slow or un-responsive DHCP server, in this case the KWS might start using 127.0.0.1 (local host) as IP address and subsequently get another IP address from the DHCP-server. The above described scenarios are now handled more gracefully.
Add users.xml with debug information to the exported log. This way an exported log will include information about all handsets in the system. For KIRK handsets even the current handset models and firmwares used will be listed. For security/privacy reasons user access codes and authentication passwords will not be included in the debug users.xml file.
Some minor performance enhancements have been made when handling Lync users.
An unnecessary XML header has been removed from Lync requests.
Release Notes
Q4 2014
31
KIRK Wireless Server 6000, Release Notes
The system MAC-address is now cached to improve sending many REGISTER requests to
Lync.
The device certificate of the unit is now displayed in the WEB-GUI Configuration | Certificates menu. The device certificate will only be displayed if the unit was produced with a device certificate. For a description of device certificates refer to section: 4.4
PCS12BC
– Q2, 2012 of this release note.
Version
The RTP handling has been made more robust by adding additional filtering to avoid treating non-RTP packets as RTP.
GUI menu and HTML handling changes for more correct html/javascript syntax and stylesheet handling of "<div id=\"background\"> tag.
On some browsers the old way of doing it would mess up rendering of some <div> tags other than the background <div> tag.
From the WEB-GUI it is possible to export user data as an CSV or XML file (Users |
Import/Export). If such a file is exported the HTTP content-type of the file is changed from application/octet-stream to "text/csv" and “text/xml” respectively.
Make RemoveIOHandler and ModifyIOHandler more robust to invalid file descriptors.
Added some more tooltips in the WEB-GUI.
Removed Features
None
Corrections
An issue with comparison of certificate validity timestamps has been addressed. This corrects a problem where the KWS claims a certificate to be expired if it expires within the current year. For example, if the current year is 2013 and the certificate expires in December
2013 the KWS will claim it expired starting from January 2013. The consequence of the bug is that the KWS will be unable to make connections via TLS for SIP or provisioning if the server presents a certificate that expires within the same year.
Fixed a problem where a media resource with CODEC card detects early media even though no early media is received. As a consequence the local ring back tone is disabled and silence is received while the remote end is alerting. This change addresses an issue reported in DECTESC-438.
Previously the KWS call statistics (WEB-GUI: Statistics | Wireless Server) could in some cases report a wrong (too high) number of active outgoing calls. This has been corrected.
The issue was reported in DECTESC-436.
Allow omitting empty AC code in provisioning users.xml. If the AC code was omitted from the
XML, an AC code of 00000000 was inserted. The correct behavior is to insert no AC code.
When receiving a REFER with additional parameters in the Refer-To header, filter these correctly before using them in an outgoing INVITE.
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KIRK Wireless Server 6000, Release Notes
Minor correction to the quick status available on the WEB-GUI at Status | General Quick status. If no response is received from the NTP server the Display NTP status will now display failed.
Remove a nonsense check that did not allow connecting the base station with RfpNo 256.
This bug only allowed for connecting 255 base stations.
Correct a typo that required a restart to make changes to sip.tcp_contact_ephemeral_port to take effect.
Configuration File Parameter Changes
File Action Parameter
config.xml Added sip.vlan_cos config.xml Added sip.media.vlan_cos
Description
This setting controls the signaling 802.1p Class-of-
Service Priority Code Point
(PCP): A 3-bit field which refers to the IEEE 802.1p priority. It indicates the frame priority level. These values can be used to prioritize different classes of traffic
(voice, video, data, etc.). The setting requires VLAN tagging to be enabled.
Values: 0 to 7
Default: 3 (critical applications).
This setting controls the RTP
802.1p Class-of-Service
Priority Code Point (PCP): A
3-bit field which refers to the
IEEE 802.1p priority. It indicates the frame priority level. These values can be used to prioritize different classes of traffic (voice, video, data, etc.). The setting requires VLAN tagging to be enabled.
Values: 0 to 7
Default: 5 (voice).
Version PCS12D_ – Q4, 2012
Added or Changed Features
The corporate phonebook has been updated based upon customer feedback.
The Corporate phonebook now supports special characters (ISO 8859-1 ISO Latin-1). This
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KIRK Wireless Server 6000, Release Notes means that it is now possible to search for e.g., Müller in the corporate phonebook.
The LDAP-searches made by the corporate phonebook no longer automatically follow references when retrieving records. This could cause some searches to return with the following error code:
"ldap_search_ext_s: ldap://<Servername>: Operations error"
The corporate phonebook now supports LDAP paged searches, it retrieves 500 records at a time.
Some customers have had issues interfacing with LDAP servers. The KWS6000 can handle up to 40.000 entries in the LDAP internally. However, some LDAP-servers encountered have a default query limit of 1000 objects per request. Previously if the LDAP user query return size was lower than the number of returned items from the LDAP server matching a search, the KWS would log the following error:
"error ldap_search_ext_s:ldap://<Server>:Size Limit exceeded" and the corporate phonebook would not work as desired. With support for LDAP paged searches, this is no longer an issue.
Previously LDAP-searches were blocking, i.e. the corporate phonebook would be inaccessible when performing a LDAP-refresh. In a setup with a slow LDAP server/slow network connection to the LDAP-server, combined with a large number of records in the
LDAP-search, this could lead to a degraded user experience with an unresponsive corporate phonebook.
In this situation typically the message log would contain the following warnings:
“msfphonebook: IOHandler 'msfphonebook' used xxxxx ms.” and
“msfphonebook: HandleIO used xxxx ms [yy I/O events].”
The corporate phonebook now accesses the LDAP server asynchronously for better performance, and a better user experience.
When a reset-to-default/reset-to-factory settings is performed, the corporate phonebook is now stopped gracefully.
The handling of certificates has been improved.
Previously, the KWS would not support certificates with an expiry date after year 2038, these certificates are now supported.
The calculation of SHA1 fingerprints for CA certificates, (available in the web-GUI at
Configuration | Certificates), could for some certificates calculate an incorrect SHA1 fingerprint, this has been corrected.
In the list of CA certificates the Subject is displayed instead of Issuer in the common name and Organization column. This is important for intermediate CA certificates. While the subject and issuer is the same for a root/top-level CA certificate they are different for an intermediate
CA certificate. Previously an intermediate CA could be mistaken for the root CA.
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KIRK Wireless Server 6000, Release Notes
The interface between the media resource and the KWS has been updated which means that the protocol version is bumped from 10 to 11.
The media resource will send a measure of the current performance load imposed by codec and voice packet handling every time the load changes. This performance measure is reported back to all connected KIRK Wireless Servers (normally one, in the case of redundancy however it will be two). The current - and max. load in percent is available for all connected Media-resources on the Web-GUI at Statistics | Media Resource.
The algorithm for choosing best media resource when allocating a new channel now uses available performance on the media resource, instead of available channels, as criteria to select the best media-resource. This is to prepare for newer generations of media-resources where the number of available channels potentially depends on the codecs used.
When handling codec-negotiation the KWS will no longer prefer to allocate a G.729 capable
Media-resource if G.711 or G.726 is given higher priority than G.729 in incoming calls.
I.e. if the KWS finds a G711 or G726 codec which both the remote end, and the KWS supports, and no G.729 codec with higher priority than the G.711 or G.726 was found, there is no need to prefer to allocate a G.729 capable Media Resource.
The web-GUI has undergone a minor update. It now has got a simpler cleaner look, with a new banner picture and new icons. Furthermore some typos have been corrected.
The media resource interface has been made more robust to faulty signals from the KWS.
Previously in some error scenarios, the KGAP could send faulty signals to the media resource. The input checking of KGAP signals has been made stricter to prevent future error scenarios like these.
The handling of incoming IM from Lync has been improved. The IM was discarded but too much processing was performed before rejecting the IM. Now the IM is rejected without further processing.
Previously in the case of a firmware update initiated by provisioning, the provisioning quick status indicator, which is located on the web-GUI in the Status | General menu, would indicate failed in the time span after a firmware update is successful but before the KWS has re-booted to make the new firmware active. This will be the case when the KWS waits to perform a reboot until the system is idle, i.e. until there are no active calls on the KWS. Now the KWS quick status will not indicate failed if the KWS after a successful firmware update is in the state waiting to perform a reboot.
Several components in the software platform have been updated to prepare for next generation hardware platforms. The next generation hardware platform will update the current IP-DECT product portfolio with new additions to the IP6000 family of products.
The framework for firmware update has been prepared for next generation Hardware platform.
The media resource codec handling is prepared for the inclusion of Software Codecs.
The platform now support sysfs style LED control.
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KIRK Wireless Server 6000, Release Notes
Handling of DTMF events has been improved. The way the KWS attempts to adjust incoming
DTMF timestamps to get them in line with the voice packets, has been improved.
The configuration daemon which handles all configuration settings has been improved.
At system start-up, the configuration daemon is allowed some extra time to start-up before proceeding. Subsequent processes rely on the configuration daemon for access to settings.
In some rare cases this was seen to give problems.
When the configuration daemon is persisting settings by writing to configuration files, this is now handled by writing a new file and making an atomic rename.
This will decrease the probability of problems, in the event of a loss of power to a KWS while persisting settings.
The Vendor string in Ms-Device-Info header (used with LYNC registrations) has been changed from Polycom Kirk to KIRK.
The Hardware and Firmware revision/PCS is now accessed using the config daemon, instead of parsing various system files.
The logging daemon which is handling logging of debug-, information-, warning- and errormessages has been improved. The internal format for messages logged to the Log Daemon has been changed to enable better cross-platform support. The change however is backwards compatible and will not break backward compatibility except for internal debug tools which have not been released to the public.
Removed Features
SIP usernames are made case-insensitive.
Until now the KWS has allowed for creating two different users which only differ in case. E.g.
SIP:[email protected] and SIP:[email protected].
A SIP username is however defined as case-insensitive, thus this has been changed. The
KWS will no longer allow for creating different users, which only differ in case.
A specific PTA reset function has been removed due to the fact that it is not used by PTA.
Corrections
An issue with call waiting has been resolved. In a Microsoft Lync setup, with call waiting enabled on the KWS, the KWS had an issue with Instant Messaging. Specifically if call waiting was enabled, and a user was in an active voice call and an incoming IM was received the KWS could restart. This issue has been resolved. This issue was introduced in firmware
PCS12C_.
The KWS did not send a final response if media resource allocation failed for an incoming call. This caused the call to hang, until a timeout or termination from the caller. Now a 503
Service Unavailable is sent. This issue was introduced in firmware PCS12C_.
For the XML-RPC events endpoint_base_stations and endpoint_status the order of the data fields “from” and “type” were interchanged and did not match the documentation. This is corrected.
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KIRK Wireless Server 6000, Release Notes
When accessing handset data via the XML-RPC method endpoint_nv_data the result is returned in an endpoint_nv_data event. This event did not include a from field to identify the handset. This is added.
A buffer overflow which potentially could cause a restart of the KWS if an endpoint_call_menu with several long items were received on the XML-RPC applicationinterface has been identified and eliminated.
The XML-RPC endpoint_call_menu() method did not truncate too long menu items correctly.
This is now corrected, so that items longer than 15 characters are truncated, and still sent to the handset.
The functionally for importing a configuration file in XML format via the GUI did not work and failed silently. The imported configuration was simply ignored. This is fixed.
In some transfer scenarios the transferor sent an extra BYE request to the transferee. This did not cause problems but was incorrect behavior. The behavior was specifically seen with
Nortel CS1000. This issue was introduced in firmware PCS12C_.
A buffer overflow in the XML-RPC function endpoint_broadcast() that could cause a restart of the KWS was identified and eliminated. The buffer overflow occurred when a broadcast text of more than 19 characters was sent.
When enabling and disabling the corporate phonebook several times, the corporate phonebook was not enabled correctly, and required a KWS reboot before it was working.
This is corrected.
In a scenario where several provisional responses are received during a semi-attended transfer the KWS previously would issue the following state/event error:
SipEndpoint
(PPid:xx) (UserName:John.Doe) (SipEndpointRecvInviteStatus) unexpected signal/SIP in state OutgoingTransfer.
This has been resolved.
A rare memory leak has been identified and eliminated.
Configuration File Parameter Changes
None.
Version PCS12C_ – Q3, 2012
Added or Changed Features
Based upon customer input and requests, some new features and capabilities have been added to the provisioning framework in the KIRK Wireless Server. Provisioning can be used to automate the process of firmware upgrade, KIRK Wireless Server configuration and user creation/deletion/modification on the KIRK Wireless Server.
The setting provisioning.config.check has been added to allow disabling of provisioning of the configuration. This is relevant, for example, when it is desirable to configure the KIRK
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Wireless Server using the WEB-GUI (which is typically done only once at installation) and use the provisioning server exclusively for controlling the creation/deletion/modification of users. Previously, this specific setup was not possible because the provisioning process required a config.xml file with the configuration before starting the provisioning process. Now it is possible to enable/disable the check for a new configuration as part of the provisioning process. The configuration setting provisioning.config.check controls whether or not the provisioning process will check for a configuration on the provisioning server. The default setting is provisioning.config.check = true.
To improve the integration with Broadsoft based provisioning systems, the provisioning of users has been made more flexible. Some customers using Broadsoft templates based provisioning have requested to provision “user templates”, which can be looked upon as users with empty usernames, and let the provisioning framework automatically fill out these
“dummy” users with data provided by the Broadsoft framework as users are added to the system over time. The username field in the users.xml file with user data previously was required to be present and unique to ensure data integrity. With this update the users.xml file is allowed to include users with empty usernames. These will be ignored by the provisioning process.
Lync Survivable Branch Appliance (SBA)
– fail over mode is now supported. In a setup with a local branch office and a WAN-link to a central Lync server, a SBA can be deployed locally which will act as Lync server if the WAN link to the remote Lync server is down. The SBA provides ongoing voice service but degraded IM and presence services to the user. This addresses DECTESC-416.
SIP handling has been re-factored to keep the platform agile and able to absorb new developments in the SIP standards.
Previously, a handover performed just when establishing a call in the small time-window while the allocation of a media resource was pending would result in calls without audio path; that is, the call signaling would succeed however the audio path would be missing. This problem was identified at a Lync site in a setup with a slow edge-server and reported in
DECTESC-412. Starting with this firmware, an incoming handover request from the handset that occurs in the time-window while a media-resource allocation is pending will be disallowed, which eliminates the problem. Furthermore, the time-window in which this can occur is drastically reduced by rewriting and optimizing the media allocation algorithm. This means that the media allocation, and thus the time-window, is now independent of the responsiveness of, for example, a Lync edge-server. This also means that the media resource protocol version is bumped.
Media handling has been improved. The handling of RTP streams has been improved with respect to handling obscure RTP scenarios and especially changes made to the media stream (RTP) during call and during call establishment. In some user scenarios the KIRK
Wireless Server will receive media streams from different sources during call setup. If the
KIRK Wireless Server receives media (RTP) with changing synchronization source identifier
(SSRC), the media handling now handles this more gracefully.
If the SSRC changes, the media handling will have to re-adjust. This may have a negative impact on the media, potentially leading to delays in the received media and possibly parts of the media missing. In pre PCS12BC versions of the firmware most changes in the received
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KIRK Wireless Server 6000, Release Notes
RTP stream resulted in the termination of one RTP session and the subsequent (rapid) establishment of a new RTP session. Starting with PCS12BC the number of these re-starts were reduced, which improved the user experience in various scenarios. While improving some scenarios it could result in degraded performance in other scenarios. These have been identified and are handled more gracefully.
Various SIP Errors/Warnings in the message log have been augmented. The internal KIRK
Wireless Server id of the handset in question (PPID) and the SIP username of the user are logged together with the error message to facilitate trouble shooting.
Allow sending status updates to a KIRK Butterfly Handset while an incoming call is established. This is required for some call waiting scenarios.
For incoming calls with late SDP signal that media negotiation is completed to the media resource when the ACK request is received. This could cause the call to fail later on in some call scenarios including KIRK Butterfly Handsets.
In some call scenarios, dialog event updates were not sent towards the other end. Dialog event updates are specified in RFC4235 INVITE-Initiated Dialog Event and used by some
IPBX’s e.g. Nortel SCS500 for Busy Lamp Function.
Major rework of the RTP statistics log message. The RTP statistics log message can be used to troubleshoot problems with missing voice packets and other network-related problems.
Some of the problems that are sometimes seen are problems with firewalls blocking different port ranges. To make debugging of this more user-friendly, the RTP statistics is augmented with the username of the involved user and the complete media path (local and remote IPaddress and port).
Furthermore the duration of the media session is now logged in milliseconds, as well as a new metric called problem packages, which is introduced and logged in packages and in percent of received packages.
Previously, the message was only logged at the termination of the call. Now statistics for the media-leg between the media resource and the base station are logged after each handover.
The old (extremely long) log message is split into two messages: an overview and a detail part.
The log level of the overview message is debug per default and info if rtp.verbose (config parameter) is true.
If the problem package percentage exceeds 5 and the session duration exceeds 5 s, the log level of the overview message is raised to notice.
Notice that calls towards endpoints which do not send any packages in the case of silence
(silence suppression) might in some cases result in “false positives” meaning that the RTP statistics might suggest RTP problems where there are none.
The error handling and checks concerning media-resource allocation has been improved.
Previously, errors could be logged concerning ICE problems even in some cases when ICE was not active, this has been changed.
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KIRK Wireless Server 6000, Release Notes
Add P-Asserted-Identity to NOTIFY send by the KIRK Wireless Server. This improves some call transfer scenarios when the KIRK Wireless Server is running as a trusted server in a
Lync setup.
P-Asserted-Identity is always added to 200 OK for PRACK. Previously, this was only the case for Lync.
When sending a RTCP sender report, cname is now sent as source description. If GRUU is available, this will be used; otherwise username@localaddress will be used.
When the KIRK Wireless Server sends a 404 Not Found, a log message with level notice “No endpoint associated with message” was logged. This is expected behavior and has been downgraded to level debug as it polluted the message log.
Make the KIRK Wireless Server manage RTP sessions for new calls. This was handled by the media resource but made it hard to decide what call a new RTP session belongs to.
If the KIRK Wireless Server is using DHCP and the DHCP server responds slowly, the KIRK
Wireless Server may start handling SIP and communicating with media resources, base stations, and so on, believing that its address is 127.0.0.1. In this case, the KIRK Wireless
Server will tell the base stations and the media resources that they should connect to address 127.0.0.1. If the media resources and base stations store this (temporary and incorrect) address, they will overwrite the (normally) correctly configured address and will no longer be able to connect to the KIRK Wireless Server. To handle this scenario, media resources and base stations will ignore a received server address if it is 127.0.0.1. This addresses DECTESC-385.
Validate configuration data before saving it in the config.xml. This is done to avoid creating invalid XML and making the system unable to read the configuration. This is done to take care of DESTESC-419 where a DHCP server sent some garbage configuration data.
When media resources and base stations are receiving their KWS address via DHCP vendor options, do not store the KWS address in the configuration before successfully connecting to the KWS. Before this fix, the address was stored in the configuration when retrieved via
DHCP. This is done to take care of DESTESC-419 where a DHCP server sent a garbage
KWS address.
Handle delayed out-of-order SIP ACK in combination with late SDP. This fixes a problem with being a member of Broadsoft hunt group, DECTESC-421.
Allow the initial incoming INVITE to the call on hold. This fixes a problem with being a member of Broadsoft hunt group, DECTESC-421.
Allow INVITE with replaces in combination with two calls. This addresses DECTESC-384 hunt group on ShoreTel. The KWS still does not support incoming REFER with two calls as seen on e.g. some ShoreTel configurations.
Previously, the KIRK Wireless Server could hang while waiting for a slow DNS-server while resolving the address of a Lync Edge server. This caused a log message with level warning about SIP TLS I/O handler using long time. This has been improved.
The firmware upgrade handling has been updated.
The mechanism for reporting the status of a firmware upgrade in progress has been
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KIRK Wireless Server 6000, Release Notes improved. Previously, the scheduling could let the flash update process monopolize the CPU, which could result in a situation where the WEB-GUI for displaying the status of a firmware upgrade would become un-responsive. As a side effect of this, central upgrade of base stations and especially media resources could stop before all base stations/media resources were updated.
Handling of central firmware upgrade of media-resources and base stations now supports the status "Identical" if an attempt is made to update a unit with the same firmware as already in the flash. Previously, a central firmware update of media resources or base stations would stop at the first unit with a firmware identical to the one used in the central update. Now, if the central firmware update encounters a unit with the same firmware as the one used for the update, the update status for the unit will be identical and the upgrade process will continue to the next unit.
For progress reporting purposes two new sub-states of a firmware update have been added:
Erasing and Writing. A central firmware update can be in the following states:
- Downloading: The firmware image is being downloaded to the unit.
- Verifying: A sanity check is performed on the firmware image file to ensure that it is a valid firmware image.
- Erasing
: The “old” firmware image is being erased.
- Writing: The new image is being written.
- Updated: The unit is updated.
- Failed: The update failed.
- Identical: The firmware image in the update request is identical to the current firmware on the unit.
The upcoming KIRK Survey handset is added to the list of known handsets. The list users
GUI table will display "KIRK Survey" as handset type if a KIRK Survey handset is registered on the system.
In different parts of the KIRK Wireless Server WEB
–GUI, a received signal strength indicator (RSSI) value is used as a measurement of the power present in a received DECT radio signal. The RSSI value for the radio chipsets used in the current IP base station generation is within the range of 0-49. Different parts of the KIRK product portfolio (base stations and handsets) use different chipsets thus it is difficult to compare RSSI values across different products. For the purpose of being able to compare the RSSI values across different chipsets, an approximation and linearization has been made. This approximation will be used across the newest generations of radio platforms. For the KIRK Wireless Server this means that the RSSI value previously given within the range of 0-49 now will be given within the range of 0-110.
The RSSI map in the WEB-GUI for base station administration has been face lifted. The
RSSI map shows any other base stations synchronized with the same system. Base stations are sorted by signal strength (RSSI). The RSSI is now displayed as pure ADConverter value
(like before 0-49) and approximated and converted to dB (0-110).
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KIRK Wireless Server 6000, Release Notes
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KIRK Wireless Server 6000, Release Notes
The sync tag in the rfps.xml file now has a rssi property (like before) and a rssi_db property
(new). The following is an example of the new format.
<sync primary="0" secondary="0" auto="false" state="primary" lost="0
" changed="0" inactive="0" source="0" rssi="44" rssi_db="100"/>
This RSSI value in dB can be used to compare with the RSSI value in the new upcoming survey handset.
The RSSI-scanner in the "hidden" ( /maintenance ) part of the base station WEB-GUI has been aligned with the new RSSI db values in the base station administration. The RSSI value is now compensated and displayed in dB.
The Logmessage issued when a base station gets synchronized/changes synchronization source displays the current RSSI value. This value is now compensated and displayed in dB.
Several minor HTML markup changes have been made to make the WEB-GUI more browser-friendly. This is to ensure the highest possible degree of compatibility with the different web-browsers on the market.
Previously, after the initial codec negotiation the KWS would stop offering codecs not supported by the other endpoint in future re-negotiations. Now the KWS will include all configured codecs in new SDP offers. This is the general case and this behavior is necessary for passing some Broadsoft test cases.
When a SIP registration for a user is completed the first time, it will be logged in the message log with level info. The log entry will specify the address of the proxy, for example, user
<user> registered at <server>. If the SIP user is registered with a Lync SBA in fail over mode, the log entry will have level notice and will indicate that the user is registered with limited functionality.
The size of the signal log in log exports and crash dump files has been increased from 128 kB to 512 kB. This is done to get more info in log exports and crash dump files.
The KIRK Wireless Server is now able to detect and handle loops or merges on a second incoming call, for example, call waiting. The KIRK Wireless Server will now respond with a
402
– Loop Detected.
Add more verbose naming of ICE I/O handlers.
Previously, the statistics module calculating traffic (call and messaging) statistics would log an entry in the debug log once a minute. The timer handling has been changed to eliminate these log entries as they did not add any real debug value and polluted the debug log with an entry each minute.
Removed Features
Previously, if a media resource would crash the, KIRK Wireless Server would attempt to reroute active media streams to other media-resources in an attempt to re-establish the audio for the call. In practice, however, it is not possible in many situations and has been removed.
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KIRK Wireless Server 6000, Release Notes
The creation and termination of an Operation & Maintenance signaling link instance is no longer logged in the debug log.
Handling of queued outgoing INVITE is removed. It was never used.
Corrections
Corrected a problem with the random number generator causing it to read from an arbitrary file descriptor. The problem could cause the WEB-GUI to become unresponsive when
HTTPS is used. This addresses DECTESC-401.
The WEB-GUI Active Calls statistics in a redundancy setup did not display the correct status for "Secure" on calls on the secondary server. Calls located on the primary server were displayed correctly, calls located on the secondary were always shown as Secure: "No" even if the calls were actually secure (encrypted). Note this was only an issue when running redundancy. This is corrected and addresses DECT-275.
Remove crash when sending an error message to a KIRK Butterfly Handset before a number is entered. This could happen in some dialing scenarios using overlap sending.
Previously, when a KIRK Wireless Server in a Lync setup was publishing the presence state of a user, a presence publication version conflict was issued when a NOTIFY was received by the KIRK Wireless Server before the OK for the SERVICE request. This caused a log notice in the message log about 409 versioning conflict. This has been corrected.
The handling of presence NOTIFY sent from a Lync has been made more robust. Previously, some scenarios could cause a restart of the KWS.
In some scenarios, if the KIRK Wireless Server received an Authentication-Info header, the
KIRK Wireless Server could restart. This has been seen on Broadsoft platforms in connection with SIP registration. This has been corrected and addresses DECTESC-423.
In some scenarios with Call Admission Control active in a Lync setup, the CAC handling could cause the KIRK Wireless Server to restart if CAC concluded that not sufficiently bandwidth was available. This has been corrected.
Previously, ICE nomination was started even in the case where the peer does not support
ICE. Although not having any impact on the user experience, this issued a log message with level error about a start NULL timer error. This has been corrected and was reported in
DECTESC-414.
Since PCS12BC, the Media type of service (TOS/Diffserv) was not tagged correctly on voice packets in RTP send from the media-resource. This has been corrected.
In connection with ICE negotiation, a log message with level critical about no registered I/O handler was sometimes issued. This had no impact on the user experience; however, it polluted the message log and has been corrected.
Fix a problem with outgoing calls on Nortel re-directing SIP server. When a proxy was configured, all out-of-dialog requests were sent through the proxy even though they were redirected by a 302 response. With this fix, the proxy is not used if a maddr is present in the new address. This bug was introduced in PCS08B_.
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KIRK Wireless Server 6000, Release Notes
If no default gateway is configured on the media resource either assigned using DHCP or statically - this caused no audio on all calls. This problem was introduced together with ICE handling in the PCS12BC release.
Fix a problem where a Cisco CUCM 7.1 conference bridge disabled DTMF and afterwards sent a re-INVITE with no SDP. This caused the KIRK Wireless Server to select payload type
255 for DTMF and made it unable to send DTMF commands to the bridge. This is a fix for
DECTESC-424.
The Reset UUID button in the redundancy configuration did not work. This is now replaced by a check box for resetting the UUID of the database.
A rare memory leak occurring when aborting a call while allocating a media resource has been identified and eliminated.
Do not log a state/event error when a key is pressed after a call is failed.
Some theoretical buffer overflows have been identified and eliminated.
Configuration File Parameter Changes
File Action Parameter
config.xml Added provisioning.config.check
Description
Controls whether or not the provisioning process should check for a configuration file
(config.xml).
Values: true/false
Default: true.
Version PCS12BC – Q2, 2012
Added or Changed Features
Microsoft Lync interoperability has been significantly enhanced with the addition of many new features.
Microsoft Lync ICE/STUN/TURN.
The Interactive Connectivity Establishment (ICE) Extensions protocol is used to establish media flow between two endpoints, even if separated by NATs/firewalls.
ICE provides a mechanism to assist media in traversing NATs without requiring the endpoints to be aware of their network topologies. ICE assists by identifying one or more transport addresses, which the two endpoints can potentially use to communicate and ICE determines which transport address is best for both endpoints to use for their media session.
With support for this protocol suite, the KIRK Wireless Server now supports calls between federated users in a Lync setup.
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KIRK Wireless Server 6000, Release Notes
The KIRK Wireless Server now supports being configured as a Lync trusted server. With the trusted server approach, it requires a minimum of configuration on the KIRK Wireless Server to get a complete and secure user authentication scheme in a Lync environment. This is the recommended configuration and replaces Kerberos as authentication method. Please consult the Lync application note for guidance on configuring the KIRK Wireless Server as a trusted server.
Lync media bypass is now supported.
Media Bypass allows for clients to communicate directly with a qualified PSTN voice gateway or qualified IP-PBX without traversing the Mediation server for media transcoding. No configuration is necessary on the KIRK Wireless Server. Whether or not media bypass is enabled is controlled by the Lync server and the settings are automatically provisioned from the Lync server to the KIRK Wireless Server.
Lync Call Admission Control (CAC)/ bandwidth management is supported.
CAC/bandwidth management allows for call control over WAN, to assure the accepted number of calls over the allocated bandwidth, and to refuse any extra calls exceeding the allowed bandwidth limit.
Support for Lync presence has been enhanced.
The KIRK Wireless Server will publish the following Lync presence states of the handsets: Ina-call, Available, Idle and Away.
If the security settings provisioned from the Lync server, are different from the security settings configured on the KIRK Wireless Server, it is now reported in the log. If possible, the local settings on the KIRK Wireless Server are changed to match the Lync settings. This is controlled by the ucPC2PCAVEncryption setting received from Lync. The following settings are available:
- "SupportEncryption"
- "RequireEncryption"
- "DoNotSupportEncryption"
If a mismatch that cannot be resolved without re-booting the KIRK Wireless Server is identified, it is logged as a Warning and no settings are changed. If a mismatch that can be resolved is identified, the local KIRK Wireless Server setting is changed to match the Lync setting and a Notice is logged.
The KIRK Wireless Server now supports Lync Scaleable SRTP (SSRTP), which is required for some types of conference calls in a Lync setup.
RTP over TCP is supported in a Lync setup.
Several new/more descriptive warning/error messages are now logged in connection with problems in the Lync configuration. Some of these will be presented in the message log; some will be presented in the handset display.
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KIRK Wireless Server 6000, Release Notes
To maximize interoperability with older ICE implementations delay sending ICE candidate checks with the USE-CANDIDATE attribute before the OK has been received from the caller.
The KIRK Wireless Server now supports reception and handling of a user ID during a DECT subscription. Some KIRK endpoints will now potentially send a user ID during subscription.
This user ID will be used to find a matching user template in the user database if the IPEI of the endpoint is not in the database.
This allows for a hosted provider to securely deploy a solution without managing IPEInumbers.
Now four ways of registering handsets are available:
1. If you want to control which specific handset is registered to a given SIP account, it is possible to create a number of SIP users each with a specific IPEI assigned. This way only a
DECT endpoint with this specific IPEI (the IPEI is globally unique), will be allowed to attempt to register as the corresponding user.
2. If you do not want to control which user or which handset is used to register to a given SIP account, it is possible to create a number of SIP users without assigning an IPEI. This way any GAP compatible DECT endpoint will be allowed to initiate a registration.
3. (NEW) If you want to control, which user is registered to a given SIP account but do not want to control which handset the user is using, it is possible to create a number of SIP users without assigning an IPEI. This way any GAP compatible DECT endpoint will be allowed to attempt a registration. If the handset supports it, the user will be able to enter a user ID during the registration process. This user ID is then used, to find the appropriate SIP user and initiate a registration attempt on the correct SIP user.
4. The final option is to enable auto creation of users. Configuration|Wireless
Server| Autocreate users. Using this option, a SIP user will be auto-created on the KIRK
Wireless Server, when a GAP-compatible handset initiates a DECT-registration. The SIP user name will be auto generated and will be IPEIXXXXXXXXXXXXX where XXXXXXXXXXX is the IPEI of the handset attempting the registration.
The media handling has been updated.
Previously fixed port numbers were used for the media stream (RTP) between the KIRK
Wireless Server/media resource and the external endpoint/IPBX. These are now scattered between base port number and base port number + 1000. This is done to accommodate ICE.
The KIRK Wireless Server now adjusts received packet duration during the session. This improves handling of RTP sessions that starts badly.
Some codec-card related handling has been improved. The previous handling could in some cases have resulted in “metallic” sound on some codec-cards.
The KIRK Wireless Server now sends Real-Time Transport Control Protocol (RTCP) receiver and sender reports.
In RTP stats log used for debug the session Id and receive timestamp jumps are now logged.
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The web-server powering the web-GUI has been updated.
Some of the information served by the web-GUI is now augmented with an expire header which indicates the number of seconds the file should be cached by the browser. This is used to prevent e.g. style sheets etc. to be sent at every request, which improves the responsiveness of the web-GUI, especially when used over a poor connection.
SRTP has been added to the description of the encryption settings in the web-GUI for clarity.
The HTML markup for displaying the log has been updated for better browser compatibility.
The Host certificate chain validity on the Configuration|Certificates GUI was not displayed correctly. This has now been corrected.
Previously, the security part of the hidden debug maintenance menu in the web-GUI was not accessible using the hidden menu itself; it is now. Furthermore, the hidden maintenance menu now displays the current Timer Queue length and the current number of registered
IOHandlers.
The P-Preferred-Identity/P-Asserted-Identity is now added to more SIP requests and responses. When the KIRK Wireless Server is configured as a trusted server, Lync requires that P-Preferred-Identity/P-Asserted-Identity is added to more SIP requests and responses.
In earlier versions of the firmware P-Preferred-Identity/P-Asserted-Identity was only added to
INVITE and 200 OK for INVITE.
The UUID and EPID of a SIP user will be changed when user information is changed. This ensures that the SIP server will not see the same UUID or EPID for different usernames when a user is renamed or the username is interchanged between two handsets.
A couple of additional host certificate formats are now supported. This includes SHA-224,
SHA-384 and SHA-512. Furthermore, support for mixed case initialization vector in PEM formatted AES encrypted key files has been added.
Added support for device certificates for e.g., provisioning. Starting with Q2 2012, the KIRK
Wireless Server will be produced with a built-in device certificate that can be used for authenticating the device towards a provision server supporting TLS client authentication.
The MAC address of the device is stored as the Common Name (CN) in the certificate. If a device certificate is present, the firmware will use it for HTTPS based provisioning and for the built-in HTTPS server.
A configuration setting for specifying the hostname of the device has been added. The hostname will be inserted into the SIP Contact and via headers. The hostname will also be published via DHCP and if the network infrastructure supports it, the device will be reachable via this hostname. The setting is available in the GUI at Configuration|General|Hostname.
If an Automatic Alarm Call Application License is loaded, Automatic Alarm Call application is enabled in the handset (if supported by the handset).
Ignore Key-presses from alerting handset while waiting for connection. This eliminates warning from the logs at some customer sites. These warnings were issued if a handset sent key-presses to the KIRK Wireless Server while alerting (seen on SWAN handsets and
Siemens handsets; standard KIRK handsets do not send these). Previously, a State/Event
Error was logged with level Warning, now a debug message is logged: "MNCC_INFO_ind received in State WaitForConnectInd, signal ignored."
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KIRK Wireless Server 6000, Release Notes
SIP Transaction cleanup time is increased from 5 to 8 minutes. This is done to delay the clean-up to after the DECT CC04 timeout (6 minutes). This eliminates an issue with missing
CANCEL and hanging call on NEC IPBX and Asterisk. The issue is reported in DECTESC-
394.
Optimized/reduced signaling for redundancy connection. This will enhance the performance in start-up and fail-over scenarios in a redundancy setup.
The handling of redundancy slave connection has been made more robust.
HandsetCompatibilityModeIE is now sent to handset at location registration & subscription.
This tells the handset to enable extra delay after a handover attempt (60 ms), which is the default setting of the handset.
For SIP over TCP and TLS do not add the ephemeral port to the Contact header. In earlier versions of the firmware, the temporary local TCP port was added to the Contact address.
This change can potentially cause problems with some SIP servers and make the KIRK
Wireless Server unable to receive incoming calls. In this case the hidden setting sip.tcp_contact_ephemeral_port can be set to true, which will revert to the previous mode of operation.
Reset TCP/TLS read and write pointers when the received buffer is deleted. This is mostly to avoid bogus SIP TLS trace messages when keep-alive is received.
Log an error if a TCP connection fails during TLS handshake - This is usually caused by the server disconnecting because of a TLS problem.
Do not log a state/event error when an active call fails in a two call scenario and
CC_RELEASE is received.
When an I/O handler is registered or removed, it is now logged with level DEBUG.
Removed Features
Remove Lync AD/Kerberos configuration from GUI. This is replaced by Trusted Servers.
Remove obsolete setting: “
Require symmetric RTP”.
The handling of a second incoming call, (only available if Call waiting is enabled) has been changed. Previously, if a second call was incoming, the user could press the hook key which would result in a termination of the active call, and on some PBXs the phone would start ringing again giving the user the possibility to accept the second incoming call. This behavior was however not consistent for different IPBXs and to some extent timing-dependant; thus this has now been changed. Now, if the user presses the hook key, the active call will be terminated, and the second call will be rejected with a Busy Here response. Standard Call waiting functionality is preserved, so it is still possible to accept, ignore or reject the second incoming call.
Corrections
If a system is configured with encryption of the internal RTP between base stations and media-resource (SRTP), the first handover after a wrap-around of an internal SRTP related counter, could potentially result in noise in the audio channel. The possibility to configure the
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KIRK Wireless Server 6000, Release Notes setting triggering this problem is only available if a security or Lync license is installed.
Disabling Base station RTP encryption (SRTP) is a workaround which eliminates the problem; however, in this release the root cause to the problem has been identified and eliminated. The wrap-around potentially causing this problem occurs approximately every 10 minutes. The problem was reported in DECTESC-395.
If an incoming INVITE with an empty Supported header is received by the KIRK Wireless
Server, it could cause problems. This has now been corrected. The scenario causing the problem was seen when connected to a BroadSoft IPBX.
Previously, the KIRK Wireless Server could crash when a 301 with no port in new contact was received in response to a REGISTER request. This has now been corrected.
This scenario could be triggered when using the Master/Slave feature of an Innovaphone
IPBX. With this feature, when a user is not found on a registrar, the registrar will reply with
301 moved permanently. The problem was reported in DECTESC-388.
Handle incoming P-Asserted-Identity with only a Tel: URI correctly. Fixes issue with missing calling party number on some Lync PSTN gateways.
A potential memory leak and a potential double free were identified in the messaging handling. These could result in issues, in a setup with a high amount of messaging traffic, e.g. in the case of an external messaging application with high traffic.
If the following specific combination of conditions were met: “TLS”: enabled, “r egister each endpoint on separate ports”: enabled
and
“TLS allow insecure”: disabled, the KIRK Wireless
Server could potentially crash when a user was disabled or deleted. This has now been corrected.
Only send NG_STATUS_EARLY when early media is present.
In Call waiting scenario terminate active call when red button is pressed.
In a scenario receiving a SIP display name enclosed by “” exceeding the allowed amount of characters, one more character than necessary was truncated from the display name. This has now been corrected.
Configuration File Parameter Changes
File Action
config.xml Added
Parameter
network.hostname
Description
Used for specifying the hostname of the device. The hostname will be inserted into the SIP Contact and via headers. The hostname will also be published via DHCP, and if the network infrastructure supports it, the device will be reachable via this hostname.
Values: A string
Default: Empty.
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File Action
config.xml Deprecated
Parameter
sip.lync.domain config.xml Deprecated sip.lync.servicename config.xml Deprecated sip.lync.password config.xml Added config.xml Added config.xml Removed sip.lync.trusted sip.tcp_contact_ephemeral_port sip.media.symmetric
Description
The domain of the Lync
Server 2010.
Default: Empty
The name of the DECT service user account.
Default: Empty
The password of the DECT service user account.
Default: Empty
The Wireless Server is configured as a trusted server on the Microsoft Lync server
Values: True, false
Default: False
Add the temporary local TCP port to the Contact address.
Values: True, false
Default: False.
Require that the other party sends RTP from the same
UDP port, as it will receive
RTP. This setting is obsolete.
Values: True, false
Default: True
Version PCS12__ – Q1, 2012
Added or Changed Features
The KWS now supports a new feature called base station auto discovery. Until now it has been necessary to configure a base station to connect it to a specific KWS. This can be done either automatically using a DHCP server and option 66 or by manually configuring the base station using the web GUI. Starting with this release, if a base station is deployed on the network and no KWS is configured on the base station, the base station will auto-discover any KWS on the same subnet as itself. The base station will connect to any KWS that responds. In the case of several KWSs on the same subnet, the first KWS which is discovered will be used.
The base station will NOT store the address of any auto-discovered KWS in its configuration.
The KWS now supports the new context sensitive handset user interface which will become available with the upcoming KIRK Butterfly Handset.
Added support for XML-RPC endpoint_nv_data() for accessing handset non-volatile memory.
Refer to XML-RPC SDK version 1.7 and relevant handset documentation for further details.
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Improved logging of TLS error messages. Now textual error message and error number is logged. This makes it easier to debug TLS failures.
User SIP passwords are stored unencrypted locally in the KWS. These passwords are not exchanged unencrypted during normal operation. However, if a local configuration file is exported from the KWS, the passwords are unencrypted.
It is now possible to store these passwords encrypted within the KWS. If a SIP authentication realm is configured, available on the GUI at Configuration | SIP | Realm, this realm will be used in the encryption of the passwords locally. This realm will be presented by the proxy when requesting authentication. Be aware that when the realm is changed, all stored passwords will be invalid.
The algorithm to be used for encrypting the passwords outside the device is md5(authusername:realm:password). As an example of a use case where the passwords might be generated outside the KWS, consider the case of provisioning the passwords from a provisioning server.
The user list on the GUI will now display the handset model names in the user list instead of the part numbers which were previously displayed.
The part numbers are still available as tool-tips on the GUI. I.e. hovering the pointer over a list entry will display the part number.
When a release is received from the handset in an incoming call, the SIP behavior is changed if the KWS is configured for Lync.
Pressing the reject (soft) button on the handset in an incoming call now results in a 603
Decline instead of a 486 Busy Here.
This is not correct according to RFC 3162 and if the KWS is not configured for Lync, a 486
Busy Here is still sent.
This addresses DECTESC-360. If a Lync call is incoming to a Lync soft client and a DECT handset sharing the same user, both will ring. Previously, pressing reject would only make the handset stop ringing. Now pressing reject will stop both clients from ringing.
The Production date is now printed on the GUI status general page. If the unit is produced before 31-08-2011, this will be indicated as production date.
If an incoming MSF with a call-back number exceeding the allowed length is received, this is logged together with the (too long) call-back number.
If a SIP UPDATE request is received by the KWS after a final response has been received, the UPDATE request is now handled. In the following some example scenarios are listed:
1. Receiving a SIP UPDATE request while performing outgoing transfer.
2. If the other end is using the SIP UPDATE request to signal incoming hold.
3. Receiving a SIP UPDATE for a waiting call.
This change eliminates issues discovered at SIPit 29 testing with various SIP user agents.
Increase XML-RPC endpoint_call_menu() item max length from 11 to 16 characters. This addresses DECT-256.
SIP Compact header names were not supported for the following SIP headers: Event, Refer-
To and Referred-By. These are now supported.
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When an INVITE is received with a
“From:” user of more than 36 characters, the DECT stack currently cannot handle it. Before it was just truncated, now it is discarded completely if too long. Thus, the (not useful incomplete) number will not be available as a number in the redialstack on the handset.
When extracting the tel URI used as P-Preferred-Identity from a Microsoft Lync publication
XML (received in a NOTIFY), make the tel URI check case-insensitive. This addresses
DECTESC-374 which reports a problem with calls through an external Gateway in a
Microsoft Lync setup.
An attempt to establish an outgoing call using the gateway resulted in an "Address incomplete" message.
Switch back to use sendonly in the SDP for signaling call hold. This was changed to inactive in the previous release. However, it was learned at SIPit 29 that the correct behavior is to use sendonly. This firmware continues to send inactive when configured for Lync.
The user experience in connection with call hold/call transfer scenarios has been updated. If a handset is being put on hold from the other end the handset will display “Held” instead of
“On hold”. If a handset is initiating a hold/transfer of the other end, the initiating handset will display Hold/Transfer.
Cleanup and re-factor APP_CC.
Cleanup and re-factor the handling of handset display messages.
When dialing second call fails; do not revert to OutgoingHold state but keep the failed call until terminated by the user.
Do not add empty CLIP information elements to GAP DECT signaling. This could mean that an incoming call would not display correct CLIP information if the display name was empty on some GAP handsets.
Blind transfer and semi-attended transfer handling has been improved.
On some SIP proxies, the user initiating a blind or semi-attended transfer is not able to make a new call before the transfer is completed.
Such an example is if Alice attempts to perform a blind or semi-attended transfer of Bob to
Carol.
1. Alice is talking to Bob.
2. Alice puts Bob on hold and dials the number of Carol.
3. Carols phone starts ringing and Alice presses the hook button to complete the transfer.
4. On some SIP proxies now Alice is not able to initiate a new call before the transfer is completed, either by Carol answering/rejecting the call or by Bob/the proxy ending the call.
5. If the transfer is pending and Alice tries to make a new call, the display on Alice
’s handset will display “Transfer Pending”; earlier the handset would display “error 002”.
DECTESC-380 Fixed the problem with re-INVITE with SAVP and no SRTP support. Remove unneeded state updates. Error messages are no longer displayed in the handset if an incoming re-INVITE fails. This addresses a problem with Innovaphone PBX. When a KWS user calls an Innovaphone voicemail from a DECT phone the phone disconnect the call when it is time to leave the message (when the recording starts) and the phone displays "Media
Failed"
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Removed Features
Removed a debug log message which was issued every time user data were changed either using provisioning or using the GUI.
Corrections
Do not set DNS from DHCP if the IP address is configured static. In earlier versions of the firmware, the static DNS settings were overridden by the settings received via DHCP.
Loading of PEM encoded certificates for TLS has been corrected. If the PEM begin tag was placed at a buffer boundary, the KWS failed to load the certificate. This is specifically relevant when loading custom CA bundles.
Only put SIPS URI in P-Preferred-Identity when using TLS and SIPS URI is enabled.
Previously, a SIPS URI was added even using TCP or UDP.
ACK for INVITE is not part of transaction. Route ACK as a new in-dialog request. This bug caused some strange behavior at SIPit 29.
Do not perform DNS SRV lookup if the proxy is configured as an IP-address or if it is configured with a port. Discovered at SIPit 29.
If the KWS receives a SIP REFER request without a Refer-To header present, the KWS will respond with 400 Bad Request. Previously, no response was sent.
Cosmetic correction to the User list part of the GUI.
When changing the IPEI of a user clear the firmware information.
SendRejectToLCE fixme fixed for more portable code.
Do not overwrite security descriptor before using it. Previously, the security descriptor had to be the last Information element in a signal, otherwise it would result in unpredictable behavior. This has been corrected.
Minor fixes that will make the TLS handling more stable.
Always clear call waiting indicator on abnormal release and clear it if one of two simultaneous calls are released. This eliminates some hanging call waiting status.
Identified Issues
In a Microsoft Lync setup ICE is sometimes necessary to establish calls between devices separated by firewalls. This is currently not supported. Support for this will be added in an upcoming release.
Configuration File Parameter Changes
File Action Parameter
config.xml Added sip.auth.realm
Description
Realm used for SIP authentication. The realm is presented by the SIP server
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File Action
config.xml Added config.xml Changed
Parameter Description
and is used for encrypting the SIP user password.
Values: A string
Default: Empty dect.ng_call_control Controls whether the new context sensitive handset user interface is enabled. If disabled, even handsets which support the contextsensitive user interface will operate in standard mode.
Values: true, false
Default: true sip.media.sdp_hold_attribute_sendonly When putting a call on hold, the KWS sends sendonly.
Configuring this setting as false makes the KWS send inactive.
Values: true, false.
Default: true
Version PCS10__ – Q4, 2011
Added or Changed Features
Handling of Calling Line Identification Presentation (CLIP) has been changed for incoming calls. If the call comes from an unknown or anonymous user, the “From URI” is suppressed and not used for CLIP display on the handset. Specifically, the following user names are suppressed: anonymous, unknown and null. This refers to DECTESC-283. Furthermore, incoming other party number is handled as digits even if parameters are added after the
“number” with a semicolon. This fixes an issue reported in DECTESC-338. When running a firmware version later than PCS 06A in a setup with a Nortel CS1000, CLIP did not work as expected. When a handset received a call, "sip@[IP number]" was displayed on the handset instead of the caller ID. This has now been changed.
Handling of CLIP in a setup with a KIRK Wireless Server connected to Microsoft Lync with an external gateway towards PSTN has been changed. In an outgoing call from a KIRK
Wireless Server through the external gateway towards PSTN, the gateway previously presented the main number as caller id on the receiving phone instead of the local number from the calling handset. The KIRK Wireless Server now extracts its telephone number from
Lync and presents it in the P-Preferred-Identity header. This is required to present the correct number when dialing to PSTN through the external gateway.
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KIRK Wireless Server 6000, Release Notes
Message Waiting Indication is now supported in a Microsoft Lync setup. I.e. if a voice mail is received it will be indicated on the handset.
If a DECT Lync service user was configured, previous releases would initiate Kerberos signaling for authentication towards the Lync server even if Lync was not enabled. Now the
KIRK Wireless Server will not start Kerberos signaling before Lync is enabled.
In some failing outgoing call scenarios, the handset waits for the user to hang-up before disconnecting the call. One of these scenarios is calling a non existing/illegal number.
Starting with this firmware the KIRK Wireless Server will disconnect the handset automatically after 10 seconds if outgoing call setup fails.
Support for sending reliable provisional responses - 100rel has been added. This will make the KIRK Wireless Server able to handle incoming PRACK requests. Reliable provisional responses on the KIRK Wireless Server are currently not used. They will come into play when support for STUN/TURN/ICE will be launched for Microsoft Lync.
Starting with this release, supported headers in outgoing SIP signaling will be added as separate headers instead of a list in a single header. This is currently required for Microsoft
Lync support.
When the KIRK Wireless Server indicates DTMF support in outgoing SDP offers, DTMF events 1-15 are indicated; previously 1-11 were indicated. The KIRK Wireless Server will actually send only 1-11 events. However CUCM 8.6.x apparently needs the offer to indicate
1-15 to be able to handle DTMF correctly.
The Call-Id generation algorithm has been rewritten. The Call-Id is used to identify a call in
SIP signaling. Previously, the Call-Id contained the IP-address of the KIRK Wireless Server.
This has been changed for privacy reasons.
The proxy port setting was previously required to be either 0 or between 1000 and 65535.
Now the setting is allowed to be 0
– 65535 to enable connection to a Microsoft Lync Edge server.
Handling of a second incoming (Call waiting) call while the user is on hold has been changed. In some PBX setups the following scenario could occur: if a handset is put on hold, and a new (Call Waiting) call is received, the second call will be indicated on the handset. If subsequently the second call is cancelled by the caller, the handset would not indicate this the right way. This behavior has now been changed.
Simple handling of 305 Use Proxy response has been added (this was required by Coral
UGW).
Handling of several scenarios in a setup with KIRK Wireless Server Redundancy has been improved.
It is now required that the backup KIRK Wireless Server has the same firmware revision and licensed features as the master KIRK Wireless Server to be able to connect to the master. In case a backup tries to connect with a different firmware version or other licensed features than the master, an error message will be logged and the backup will be rejected.
Furthermore, the logic to control the connection between the master and backup KIRK
Wireless Server has been improved.
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When used in a setup with KIRK Wireless Server redundancy, the storage of the primary and backup KIRK Wireless Server addresses has been improved. If major configuration changes were made to the redundancy setup e.g. if the hostnames/IP-addresses of the primary and backup KIRK Wireless Server were changed and the roles were swapped i.e. the former primary KIRK Wireless Server was re-configured to be the secondary and vice versa, the media resource could lose the correct hostnames/IP-addresses of the KIRK Wireless Server and a manual reconfiguration was required to make the media resource connect again.
When acting as a media resource, set TOS correctly when connecting to the backup KIRK
Wireless Server in a redundancy setup.
The base station administration page on the WEB GUI has been improved:
Previously a green icon indicated that the base station was synchronized to the primary synchronization source, and a yellow icon indicated that the base station was synchronized to the secondary synchronization source. The yellow icon has been removed and a green icon now indicates that the base station is synchronized.
Furthermore, the Sync Changed column has been removed from the GUI. This is done to keep the GUI as clean and simple as possible. The number of times the synchronization source has changed does not affect the performance of the system and therefore this information is not relevant.
The naming of synchronization sources has been made consistent. Now the secondary/alternative synchronization source is consistently denoted secondary. The current sync source is now printed in parentheses in the Prim/Sec column. Previously the current synchronization status was displayed even when the base station was disconnected. This was misleading and has been changed.
If a base station firmware is outdated this is now indicated by a yellow icon, (if a media resource firmware is outdated this is now also indicated by a yellow icon).
More information is logged regarding DECT subscription of handsets. It is now logged with level info when a handset attempts to subscribe to the system. Furthermore it is logged if a subscription attempt fails because of timer expiry.
When a handsets which is not in the subscription database contacts the system (e.g. performs a location registration), the KIRK Wireless Server will request the handset to delete the subscription. This is now logged with level notice.
If a Location Registration is attempted with corrupt or missing subscription data, the subscription will be terminated on the handset. This is now logged.
If a call attempt with missing or corrupt subscription data is made, it will cause the DECT subscription to be terminated. This is now logged. Furthermore, some additional textual representations of error codes are added.
Handling of certificates has been made more user friendly. When a Certificate Authority (CA) bundle is loaded into the KIRK Wireless Server and an invalid or unsupported CA certificate is encountered, is now logged. Furthermore "(Certificate no:## in file)" is logged in order to
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KIRK Wireless Server 6000, Release Notes locate the problematic certificate easier. The handling of invalid or unsupported CA certificates is improved. Invalid or unsupported certificates are listed in the GUI.
For SIP over TLS, verify the server certificate against the proxy name instead of the request
URI. This is required to make calls to another domain via a proxy. Such an example is federated calls in Microsoft Lync.
Correctly handle TLS connection shutdown from the far end. Previously, some error scenarios with TLS connections have been seen to cause a high CPU-load. This could have been influencing SIP call handling. Furthermore, this may explain some problems encountered with HTTPS connections on some browsers.
Use local media attributes to determine what RTP directions to enable.
When putting a call on hold, send inactive instead of sendonly in the SDP. This is more correct because we do not send music-on-hold. This is configurable via sip.media.sdp_hold_attribute_sendonly which defaults to false. If this setting is true, old behavior is restored.
The RTP profile, including crypto parameters is now persistent for the duration of a call. If e.g. SRTP is enabled and optional, the resulting RTP profile after negotiation with the other end is used throughout the call. Previously, re-INVITE scenarios e.g. when setting on hold would initiate a new negotiation and result in e.g. new crypto parameters. Some endpoints require the negotiated RTP profile to be persistent for the duration of the call. Such an example is Microsoft Lync clients.
Log messages concerning SIP errors have been improved. Additional information about the state of the call as well as the state of the SIP endpoint has been added. Furthermore, the remote IP address on SIP connection failures is now logged for easier debugging of SIP related error scenarios.
In the abnormal call release statistics more release codes have now a textual representation.
E.g. "Authentication failed".
A new timer concept has been added which drastically reduces the amount of timeout entries in the debug log.
When an unexpected event is received in response to a SIP SUBSCRIBE request, the name of the event is now logged.
The log daemon which collects messages and signaling for logging and debugging purposes has been improved. When sending a log entry to listening clients, the log daemon no longer gives up on all clients if sending to one client fails.
The default Certificate Authority (CA) bundle with trusted CAs has been updated. It was found that a Certificate Authority (CA) issued fraudulent HTTPS certificates. The compromised DigiNotar certificate has been removed, rendering any HTTPS certificates signed by that CA as untrusted.
For the central phonebook, the LDAP search timeout has been increased from 5 to 15 seconds in order to wait for slow LDAP servers.
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Removed Features
None.
Corrections
Sending DTMF signaling (key-presses) via SIP INFO in an early dialog could cause the
KGAP to restart. This has now been corrected. This refers to DECTESC-340.
A problem reported in DECTESC-323 has been fixed. The problem was caused by a call with a Cisco endpoint supporting both audio and video. The SDP answer received in the last ACK contained audio and video but with two different IP addresses in the c lines. The KIRK
Wireless Server selected the audio but it selected the IP address from the video; as a result the KIRK Wireless Server sent RTP to the video address. This has now been fixed.
Set local port for incoming SIP TCP connections. This makes the KIRK Wireless Server to use the correct source port when responding to a request sent directly to the KIRK Wireless
Server, by passing the proxy.
When DNS SRV and Use SIPS URI are enabled, the KIRK Wireless Server now requests
SIPS from DNS SRV.
If an incoming call is initially put on hold using IP address 0.0.0.0, the media is now correctly connected. This fixes DECTESC-344: no audio in incoming group call on Broadsoft R17.
If a system running redundancy has more than 12 active calls and an active call list is requested from the WEB-GUI, a memory error will occur. This has now been fixed.
Receiving too many A records from the DNS server previously triggered a buffer overflow.
This has now been corrected. The first 10 A records are now handled and the rest are ignored.
Some potential buffer overflows have been eliminated.
Some potentially dangerous faulty error message formatting has been eliminated.
Offset error in SRTP replay check has now been fixed.
Always update Connection->General.LifeTime when sending TCP messages.
If SIP signaling is received with an incoming Refer-To address without a user part, the KGAP could restart. As an example (probably rather theoretical) this can occur when transferring an active call to voicemail on Microsoft Lync. This has now been corrected.
Previously, the KGAP could restart if a certificate with no names was received in a SIP TLS call. This has now been corrected.
Some small memory leaks in connection with handling illegal certificates have been removed.
An unimportant state-event error which was issued when receiving a re-INVITE without SDP has been removed.
An UPnP related issue has been fixed. A multicast packet missing the ST header could cause the administration process to restart.
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If auto creating a user failed (e.g. because of a wrong access code) the temporary user was not deleted. This has now been corrected.
If auto assigning an IPEI to a user failed (e.g. because of a wrong access code), the temporary IPEI was not cleared. This has now been corrected.
Identified Issues
In a Microsoft Lync setup ICE is sometimes necessary to establish calls between devices separated by firewalls. This is currently not supported. Support for this will be added in an upcoming release.
Configuration File Parameter Changes
File Action
config.xml Added
Parameter Description
sip.media.sdp_hold_attribute_sendonly When putting a call on hold the KIRK Wireless Server sends inactive (which is the more correct) instead of sendonly (which is old behavior). Configuring this setting as true restores old behavior.
Values: true, false.
Default: false
Version PCS09__ – Q3, 2011
No significant changes since PCS08B_. This is a GA version of the PCS08B_ firmware.
Version PCS08B_ – Q2, 2011
Added or Changed Features
Microsoft Lync interoperability added.
In scenarios where it is preferred that the user sends all SIP requests to the same proxy, the
IP address of the proxy is now stored instead of the hostname when an OK is received for a
REGISTER request. This will ensure that when a user is registered on a proxy it will use that proxy for all outgoing signaling, even in the case of a proxy name which resolves to multiple addresses.
The above described behavior is enabled in the following scenarios:
○
If “send_to_current_registrar” is enabled. This is typically used in Cisco Unified Call
Manager scenarios.
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○ If Microsoft Lync is enabled.
With the above described behavior it is possible to have a setup with load sharing, while at the same time preventing that users potentially move from one proxy to another for every call.
SIP proxy-handling and routing of SIP messages has been improved.
If “dns method” is configured to DNS-SRV the KWS will attempt to resolve pre-configured proxies via DNS-SRV. If this fails they will still fall-back to A-records.
Furthermore the KWS will no longer add any pre-configured proxy in a Route header for outbound requests outside a dialog.
DNS A-records with multiple addresses are now handled and used like prioritized SRV records. Earlier if a DNS A-records answer with multiple addresses was received only the first address was used.
If the KWS receives an INVITE or an UPDATE with an unknown Content-Type, it will
Respond with “415 Unsupported Media Type”. This is a prerequisite for ad-hoc conferencing on Microsoft Lync.
The KWS will no longer terminate the active call if a BYE is received for the held call while a
REFER is pending for the active call. This resolves a problem introduced in PCS08__ which is known to cause problems with attended call transfer on Siemens OpenScape.
The product ID is now displayed in the Status General part of the WEB GUI for easy reference. Furthermore the product ID is included in a log dump (Status|Export Logs).
The KWS is now able to handle the release reason “MsfHandsetInCharger”. This release reason is only relevant for applications which enable this feature in the handset. The feature is supported by the KIRK 60- and 70- Handset Series.
The handling of SIP SUBSCRIBE requests has been re-factored to enable a more robust handling of SUBSCRIBE requests.
A warning will be issued to the Message Log if a DECT signal which is too long is sent to the protocol layers.
The internal messaging protocol between the KWS and media resources is rewritten to make future protocol updates easier.
An exponential back-off timer is now used for all kinds of REGISTER failures.
Earlier if a SIP REGISTER failed due to a SIP error response e.g. 404 not found, the back-off timer was not used, now it is.
The maximum allowed length of a SIP message is changed to 128kB on KWS6000 and 32kB on KWS300. This allows large presence XML documents from MS Lync to be handled.
Messaging handling is changed. The KWS can now handle 24 character call back numbers and 72 characters text messages. Furthermore several potential buffer overflows were eliminated.
Previously the reception of the SIP status responses: “181 Call Is Being Forwarded” and “199
Early Dialog Ter minated” resulted in an entry in the message log: “INVITE unexpected
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” with level NOTICE. This is no longer the case as these responses are perfectly legal and not unexpected.
If the KWS receives a SIP URI as callback number (a call back number starting with sip@ or sips@), the characters “p” and “s” will no longer be converted to 0x05 a pause-digit.
When TLS domain verification fails the allowed domains and the failed domain are logged to the message log. This makes it easier to find and correct errors in the TLS configuration.
An info message about re-registration is no longer logged every time a TCP connection is destroyed.
If a packet capture is stopped or started it is now logged in the message log with level INFO.
Furthermore it is logged if the packet capture is restarted due to overflow.
The size of media resource messages is limited before they are sent.
SIP TLS pcap trace messages are truncated to fit into UDP datagrams.
If the “capture everything” setting is used in packet traces, decrypted SIP messages are included in the trace. Furthermore unencrypted SIP messages were previously potentially logged twice, this is corrected.
Removed Features
None.
Corrections
If the proxy uses strict routing the KWS would crash (this behavior was introduced in
PCS07__). Support for strict routing is now restored. Specifically this addresses problems experienced when using a Toshiba CIX200 running software version MT4.1 together with
KWS firmware PCS07 and PCS08. This fix addresses DECTESC-279.
A SIP TCP buffer overflow crash and memory leak has been identified and corrected.
If a reset device to default procedure is performed any custom ca-bundle installed will be removed and replaced by the default bundle. This was not the case in earlier releases.
Changed handling of a SIP NOTIFY with a refer event. This results in a more correct call teardown in call transfer scenarios on some versions of Asterisk.
For SRTP remember crypto attribute tag received in offer. This fixes a problem where the
KWS used the wrong encryption key when more keys were available.
Identified Issues
In a Microsoft Lync setup ICE is sometimes necessary to establish calls between devices separated by firewalls. This is currently not supported. Support for this will be added in an upcoming release.
Issues have been identified in a Microsoft Lync setup with an external gateway towards
PSTN or GSM.
In an outgoing call from a KWS through a gateway to e.g. a PSTN phone, the gateway may
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KIRK Wireless Server 6000, Release Notes present the main number as caller id on the receiving phone instead of the local number from the calling handset.
Presence status for a handset on a KWS will not be correct unless the Lync account for the handset has been activated at least one time from a presence enabled client, e.g. a Microsoft
Lync client.
Message Waiting Indication is currently not supported on Microsoft Lync.
Configuration File Parameter Changes
File Action Parameter
config.xml Added sip.lync.enable config.xml Added config.xml Added config.xml Added sip.lync.domain sip.lync.servicename sip.lync.password
Description
Enable Lync Server 2010 support.
Values: true, false.
Default: false
The domain of the Lync Server
2010.
Default: Empty
The name of the DECT service user account.
Default: Empty
The password of the DECT service user account.
Default: Empty
Version PCS08__ – Q2, 2011
Added or Changed Features
Starting with PCS08__ the firmware is prepared for the Security Package license. If a
Security Package license is installed, various security enhancing features become available.
Encryption of external as well as internal media according to RFC 3711 (Secure RTP or
SRTP) is possible. External media is the media stream between the KWS/media resource and the external endpoint/PBX. Internal media is the media stream between the KWS/media resource and the base station. This addresses DECT-143.
Encryption of external media
External SRTP handling is supported in optional as well as required mode.
Configuration of external SRTP is located in Configuration | SIP Media.
If ‘enabled’, SRTP is supported and optional, and it must be negotiated with the remote endpoint. If ‘enabled and required’, the use of SRTP is mandatory, and if negotiation of SRTP with the other end is unsuccessful, call establishment will be aborted.
Handling of RFC 4568 SRTP lifetime key parameter and Master Key Index parameter in SDP offers are configurable.
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If external SRTP is enabled, the number of available voice channels on a KWS/media resource is reduced from 32 to 16, (if a codec card is used from 24 to 16).
Encryption of internal media
Configuration of internal SRTP is located in Configuration | Wireless Server.
The “Enable base station RTP encryption” setting will enforce the use of secure RTP for base station audio connections.
If internal SRTP is enabled, the number of available voice channels on each base station is reduced from 12 to 6.
Handset display handling during SIP calls has been improved with respect to displaying other party username and display name.
P-Asserted-Identity or Remote-Party-ID, INVITE and INVITE responses are used to update the handset display with other party username and display name. Specifically, this makes the handset update the display correctly during call transfer and call forwarding on several SIP servers including Cisco Unified Call Manager. Furthermore, this makes the display name of the other party update correctly in connection with group call pickup on a 3CX PBX.
Putting calls on hold before performing an attended transfer.
In previous versions of the firmware only one of the calls were put on hold. Now both calls are put on hold. This makes attended transfer work on some configurations of Cisco Unified
Call Manager Express.
Allow INVITE with a Replaces header for existing incoming early dialog.
This makes semi-attended transfer work with Polycom SoundPoint and some SIP proxies, for example IPFx and OpenSer. This addresses DECTESC-266.
Media resources and base stations are allowed to connect to a KWS even in the case of incompatible protocol versions on the KWS and the media resource/base stations. This means that in future/upcoming releases it will be possible to update the firmware on base stations and media resources directly from the KWS even in the case where the KWS was updated to a newer (and potentially incompatible) firmware version and rebooted before the base stations/media resources.
Support for VLAN tagging according to IEEE 802.1Q has been added.
VLAN tagging is statically configured through the GUI or via provisioning. This addresses
DECT-43.
Quick status has been added by popular demand. The reasoning behind quick status is a wish to make it even easier to get an overview of the health of the system. This is accomplished by adding a status summary of different vital system components to the first webpage you see when logging in to the system. To get a more detailed status it is still necessary to go into the specific parts of the GUI.
Quick status displays a status overview of the following system components:
SIP:
The SIP status is OK if all enabled SIP users are registered to the SIP server.
KWS redundancy:
KWS redundancy is OK if the connection to the redundancy peer is OK. The KWS redundancy status is not shown if redundancy is not installed on the KWS.
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Base stations:
Base station status is OK if no synchronization loops are detected, auto synchronization is disabled for all base stations and all enabled base stations are connected and synchronized.
Media resources:
Media resource status is OK if all enabled media resources are connected and at least one channel is available.
Provisioning:
Provisioning status is OK if the latest communication with the provisioning server was successful.
NTP:
NTP status is OK if the latest communication with the NTP server was successful.
SIP user provisioning changed.
Changing a SIP user setting using provisioning will have immediate impact. As an example changing the SIP display name or the SIP authentication password will be effective immediately without any issues. Some deployment configurations utilize this, e.g. for changing the display name automatically, for instance if a phone is shared between different users. One notable exception to the above is the SIP username of the user. The username is used as the key in the users.xml and therefore, changing the username was previously handled by deleting the old user and creating a new one. A result of this was that the handset was unsubscribed on the DECT side when the username was changed. Changing other fields would not cause this. Furthermore, changing the username through the GUI (and not using provisioning) would not cause this to happen.
Now the user handling is rewritten to avoid deletion of users when their usernames are changed. For this to work, the user must have an IPEI in the users.xml file.
With this new feature, it is now possible to implement a “hot-desking” feature where even the username used to call the user can change automatically controlled by provisioning. One use case is to have a device pool of handsets; when a user signs in at a shift, the user grabs any handset, calls a special number and enters an employee id. Subsequently, the provisioning framework configures the phone with the username, passwords, display name etc. without any manual interaction.
This change addresses DECTESC-214.
HTTP redirect in connection with provisioning allowed. If a provisioning server redirects a provisioning request with a 3xx redirection, this is now supported by the KWS.
Backslash in SIP authentication user allowed. This is required for NTLM where the authentication user is typically entered as: domain\user.
If an error response is received for a SIP transaction, the reason text, which in some cases is presented to the handset and logged in the message log, is now handled differently.
Previously a reason text was derived from the status code, i.e. a 488 would always be converted to "Not Acceptable Here". Now the reason phrase is retrieved directly from the received SIP message. As an example an outgoing call attempt on a Microsoft Lync may result in a SIP 488 with a text like: "Encryption levels not compatible". Previously the message log would say: “Not acceptable here”, now it will say: "Encryption levels not compatible".
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License key handling has been updated. Previously 7 licenses were allowed, now up to 15 licenses can be loaded.
If TLS is used for SIP signaling and a TLS handshake failure is experienced, it will be logged as an error. Previously this was logged as an info.
When using TLS for SIP signaling, the SIP signaling is encrypted. While increasing the security of the system, this also makes it very difficult to debug SIP signaling. The embedded packet capture functionality in the KWS now supports export of a decrypted version of the
SIP signaling. This feature is controlled by the “Capture SIPS (decrypted)” checkbox on the
KWS GUI in the packet capture menu.
If a 301 Moved Permanently is received by the KWS as a response to a REGISTER requests, the KWS will now handle this and resend the REGISTER to the new destination.
If the SIP setting “Send all messages to current registrar” (controlled by sip.send_to_current_registrar ) is enabled, 301 Move Permanently will not work.
When a SIP INFO request is received, the KWS responds with 200 OK. In previous firmware revisions, the KWS responded with 501 Not implemented. This addresses a problem reported in [DECTESC-254]. In a setup with an Aastra 5000 PBX, the PBX apparently uses
SIP INFO for keep-alive signaling. If the Aastra did not receive a 200 OK, the call was terminated.
Support of reception of RTP packets with CSRCs.
CSRCs are used for Contributing Sources. Previously the reception of RTP packets with
CSRCs could potentially lead to faulty decoding of RTP packets and resulting noise/cracks in sound. This has been fixed.
RTP stream handling improved.
Previously an RTP stream would be reset if a call was put on hold, and a new stream would start when the call was resumed. Now the RTP stream is suspended when put on hold and resumed when the call is taken off hold. This eliminates some potential RTP synchronization issues which could lead to noise during on/off hold.
The logic for handling RTP stream synchronization between local and remote ends is improved with regard to handling the scenario where the other end uses silence suppression.
In an outgoing call the KWS now waits for RTP in early media before connecting the endpoint. This addresses the scenario where the other end signals early media e.g. for generating a ring-back tone and the early media has not been received. If the early media is not received, we now generate a local ring back tone until the actual RTP is received.
The KWS now supports sending DTMF tones via rfc2833 in outgoing calls before the other end has answered the call. In some situations the user is required to enter a pin code before the call establishment can be completed. This is now handled by allowing DTMF transmission while early media is being received, and the 200 OK response is not yet received.
This addresses DECTESC-263. The specific scenario is a user which has signed up for a carrier-based user authorization for long distance calls. When one of the users calls a long distance number, he will hear a beep and will then have to enter the
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KIRK Wireless Server 6000, Release Notes authorization code to complete the call. In the scenario in question, the carrier side does not answer the call until the authorization code is validated.
Corporate phonebook LDAP access has been improved. The allowed length of ldap_bind_user is increased from 64 to 256. This addresses DECTESC-268 which involves an LDAP bind user of 72 characters.
When new base stations or media resources are rejected due to actively disallowing them
(controlled by security.allow_new_media_resource & security.allow_new_rfp), it is explicitly written in the message log that this is the cause of the rejection.
Minor changes to the Kirk Wireless Server GUI.
The following settings are now available from the web GUI on the KWS under
Configuration|SIP general:
- Enable Globally Routable User Agent URIs GRUU
- Use SIPS URI
- TLS allow insecure
The base station configuration menu on the KIRK Wireless Server GUI which is available from Administration | Base station has been updated. Changing some of the fields in this menu would automatically reboot the base station. This is now emphasized.
The Configuration | Media Resource menu on a KIRK Wireless Server applies to the internal media resource embedded in the KWS, this is now emphasized.
Furthermore, hints are added to the media resource and base station central firmware update.
Theoretically, the DECT-slot handling can go out of synchronization with the handset because the KWS “misses” a slot. If this situation occurs, it will now be automatically detected, logged as an error in the KWS and repaired. Previously, if this happened, the handsets would no longer be able to make and receive calls until the KWS was restarted.
The situation has been seen to occur in a scenario with a very high load of multicast of large packets to the base station.
Removed Features
When an IO handler is removed while IO is pending, it is recorded in the message log.
Previously this was logged with level critical but is now decreased to debug. Specifically, this can happen with packet capture in admin.
Corrections
Previously, if a SIP From header without username was received, it would in some cases result in a restart of the KWS and subsequent loss of active calls. This was reported in
DECTESC-257 which describes a scenario with an incoming anonymous call on a Cisco
Unified Call Manager. This has been corrected.
XML-RPC default password problem introduced in PCS07__ is fixed. If the default password was never changed or saved, the XML-RPC interface in firmware PCS07__ would not allow login.
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Allow SIPS URI in Refer-To header of REFER. Previously this was not handled correctly and would sometimes cause call transfers to fail (e.g. on Kameilio) if SIP over TLS was used.
When using HTTPS for provisioning, the HTTPS connection in some cases was not established correctly due to a bug in the TLS handling. This has been corrected.
If a TLS connection (either for provisioning or for SIP over TLS) was closed by the other end during handshake, this would in some cases cause a restart of the administration or the call handling process. This has been corrected.
Port numbers are no longer inserted on GRUUs. This fixes some potential routing problems when GRUU is used.
Previously the following error message was logged in some cases: “Unknown configuration key, rtp.* setting changed.” This log message was not an error and has been removed.
Previously a reboot was required if either of the settings: sip.media.sdp_answer_single and/or sip.media.sdp_answer_with_preferred were disabled. This has been corrected.
Corrected handling of MTU (network.mtu). In earlier versions of the firmware, the specified MTU was ignored and the MTU was always set to 1500 bytes. This has been corrected.
Configuration File Parameter Changes
File Action Parameter
config.xml Added network.vlan config.xml Added config.xml Added security.srtp_rfp sip.media.srtp.enable
Description
VLAN Identifier (VID) according to IEEE 802.1Q specifying the
VLAN to which the device belongs. 4094 different VLANs are supported.
Values: 1-4094
Default: Empty.
Enabling this setting will enforce the use of secure RTP for base station audio connections.
If internal SRTP is enabled, the number of available voice channels on each base station is reduced from 12 to 6.
Values: true/false
Default: false.
If enabled, external SRTP is supported and optional. It must be negotiated with the remote endpoint.
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File Action
config.xml Added config.xml Added config.xml Added
Parameter
sip.media.srtp.required sip.media.srtp.lifetime sip.media.srtp.mki
Description
If external SRTP is enabled, the number of available voice channels on a KWS/media resource is reduced from 32 to
16, (if a codec card is used from 24 to 16).
Values: true/false
Default: false.
If enabled, the use of SRTP is required. If negotiation of
SRTP with the other end is unsuccessful, call establishment is aborted.
Values: true/false
Default: false.
Handling of RFC 4568 SRTP lifetime key parameter in SDP offers.
Values: true/false
Default: false.
Handling of RFC 4568 SRTP
Master Key Index parameter in
SDP offers.
Values: true/false
Default: false.
Version PCS07__ – Q1, 2011
Added or Changed Features
Support for using TCP as transport protocol for SIP signaling has been added.
Support for using TLS as transport protocol for SIP signaling has been added. Be aware that
TLS is supported only for outbound connections. Using TLS will allow for encryption of SIP call signaling. If TLS is enabled, UDP and TCP connections will be disabled by default to increase security.
Provisioning using HTTPS is now supported. With the possibility to use HTTPS for provisioning of users and configuration data, it is now possible to increase the level of security with regard to remote management of KWS solutions.
Added support for certificate handling. A certificate is required to be able to use HTTPS provisioning or SIP over TLS. The KWS is delivered with a Certificate Authority (CA) bundle
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KIRK Wireless Server 6000, Release Notes with common Certificate Authorities. This means that the KWS will accept certificates issued by for example Verisign out-of-the-box. In addition to the CA-bundle the GUI allows for installing a local CA certificate bundle if a certificate is generated by a local authority (e.g. a service provider or the local IT department). A certificate bundle in PEM-format may be imported.
The flash update process has been improved. It will no longer update the firmware if the new firmware is identical to the current firmware.
Trigger dumps used to persist debug information in case of a fatal error have been rewritten.
Previously the latest 10 trigger dumps were stored in the flash, but now only the first trigger dump made since last boot will be stored. This is to eliminate the possibility of excessive flash wear in the case of a repeated error scenario.
Port numbers for separate, individual SIP signaling ports have been changed. Previously, port number 5060
– (5060 + number of users) was used. Now, port numbers 15061 – (15061
+ number of users) is used.
This is relevant only when the KWS is configured to use separate individual SIP signaling ports.
Configuration key sip.proxy.transport has been replaced by the keys sip.transport and sip.dnsmethod. That way the SIP transport and the DNS method are separate making it easier to understand how these are configured. The KWS still understands the key sip.proxy.transport, but it is recommended to use sip.transport and sip.dnsmethod.
When the KWS requests to put the remote end on hold, the KWS sends an INVITE with a
SDP message indicating that the call must be put on hold. Previously this SDP had the media attribute
‘sendonly’, and the IP address of the connection was set to 0.0.0.0. Now only the media attribute
‘sendonly’ is used to signal that the call is on hold. This change has been made because some SIP implementations do not allow the 0.0.0.0 IP address. Setting sip.media.sdp_hold_null_connection=true returns to the previous behavior (not RFC compliant).
Globally Routable User Agent URI (RFC5627) support has been implemented. A Globally
Routable UA URI (GRUU) is an URI which routes to a specific UA instance. If enabled, a
GRUU will be obtained from a server and communicated to a peer within a SIP dialog. With
GRUU support the KWS can handle more advanced transfer scenarios, provided it is supported by the SIP server.
An Universal Unique ID, UUID, has been added to the user data. The UUID is used for
GRUU and is generated automatically by the KWS.
user=phone has been added to header in outgoing SIP signaling. This is required by e.g.
Microsoft OCS.
Support added for SIP URIs in called/calling party numbers. If for example an incoming call with calling party number consisting of a SIP URI (e.g. sip:[email protected]
) is received, the SIP URI can be stored in the redial-stack in the handset. To utilize this, it needs to be supported in the DECT handsets. For KIRK 50-, 60- and 70-Handset Series this will be supported starting with the following releases:
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Handset Firmware version
5020- and 5040-Handset Series
6020- and 6040-Handset Series
PCS08Na (or newer) released 2011-Q1
PCS07Ea (or newer) released 2011-Q1
7010-, 7020- and 7040-Handset
Series
PCS07Ea (or newer) released 2011-Q1
Support P-Preferred-Identity and P-Asserted-Identity headers (RFC3325). These headers allow trusted parties to assert the identity of authenticated users.
Sending BYE immediately after REFER has been made configurable. During a call transfer the existing SIP dialog can be terminated by either the transferor or the transferee. Per default the KWS will terminate the dialog with a BYE request when acting as a transferor. If sip.send_bye_with_refer is set to false, the KWS will not send BYE when acting as transferor but rely on the transferee to send the BYE.
Added support for SDP message sessions. Now SIP-servers/endpoints which use an INVITE to establish an instant-messaging session are supported by the KWS. This can be used for instant messaging towards the DECT endpoints on the KWS.
Added option for ignoring the version information in incoming SDP received from remote endpoints. The default setting is sip.media.sdp_ignore_version=false. With the default setting, the version information in the incoming SDP will be honored. If the version is not changed, any change in incoming SDP will be ignored. This option can be controlled via the
GUI on the Configuration|SIP page and through the setting: sip.media.sdp_ignore_version. If the other end changes the SDP without changing the SDP version, this setting should be true
(not RFC compliant).
When logging an abnormal call release, add a textual interpretation of the error code (if available). This addresses a feature request reported in DECTESC-236.
If the process of allocating a voice channel on a media resource for a new call fails, the reason for the failed allocation is now logged, e.g. no media resource connected or no free channels.
Logging of failure to allocate a voice channel on a media resource is demoted from
CRITICAL to NOTICE.
Status/debug information with regard to RTP handling has been added. When a RTP session is destroyed/closed, the current length of the RTP queue can now be logged.
Removed Features
None.
Corrections
If the configuration setting media_resource.enabled is changed, the local media resource will be stopped if the setting is false. All other values will result in the start of the media resource, e.g. true or empty (default). This solves DECTESC-226.
Release Notes
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An error in reboot-on-idle functionality has been corrected. Since provisioning is using the reboot-on-idle functionality as of PCS06A_, this correction will make provisioning able to reboot the device.
If the KWS receives a 423 (Interval Too Brief) response to a SIP REGISTER, it will now honor the expiration interval within the Min-Expires header field of the 423 response. This means that subsequent SIP REGISTER requests will be sent with an interval as specified by the registrar even if this interval is greater than the registration expire setting of the KWS.
DNS SRV priorities with the value 0 are now allowed. Previously the value 0 was regarded as no priority.
The negotiated ptime while sending DTMF via RFC2833 is respected.
The free (12/30) users are not lost when only a feature license is loaded.
Monotonic clock is used within SIP transaction timer handling. This means that SIP transactions will not be affected by “jumps” in local time, e.g. due to NTP corrections to the time.
CSV file import of users has been fixed, line ending is handled correctly. Furthermore, the line number in the following error message has been corrected:
"CSV import line # - Wrong number of fields expected #".
IPEI is printed correctly on auto created users.
Configuration File Parameter Changes
File Action Parameter
config.xml Added sip.media.sdp_ignore_version config.xml Added sip.media.sdp_hold_null_connection config.xml Deprecated sip.proxy.transport
Description
Specifies whether to ignore the version information in incoming
SDP received from remote endpoints.
Values: true/false.
Default: false.
If this setting is true, the
KWS will revert to the old way of signaling a hold.
Values: true/false.
Default: false.
Deprecated. In release
PCS07__, this setting is replaced by sip.transport
& sip.dnsmethod.
The KWS still understands this setting, but the new settings
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File Action
config.xml Added config.xml Added config.xml Added
Parameter
sip.dnsmethod sip.transport sip.gruu config.xml Added sip.rfc3325
Release Notes
Q4 2014
Description
should be used.
Specifies the DNS method used to resolve host names for SIP requests.
Values: arecord/ dnssrv. arecord: Use simple DNS
A records to resolve host names.
Basically A records are used to translate a hostname to an IPaddress. dnssrv: Use DNS SRV records to determine host addresses. Refer to
RFC3263. DNS SRV records can be used to specify multiple servers with different priorities and/or multiple servers for load-balancing.
Default: arecord.
Specifies the transport mechanism used for SIP requests.
Values: UDP, TCP, TLS.
Default: UDP.
Specifies the use of
Globally Routable UA URI
(GRUU) which is an URI that routes to a specific
UA instance. If enabled, a
GRUU will be obtained from a server and communicated to a peer within a SIP dialog.
Values: true/false
Default: true.
This setting controls support of RFC3325 P-
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File Action Parameter
config.xml Added sip.send_bye_with_refer config.xml Added config.xml Added
Release Notes
Q4 2014 sip.use_sips_uri sip. tls_allow_insecure
Description
Asserted-Identity and P-
Preferred-Identity headers. These headers allow trusted parties to assert the identity of authenticated users.
Values: true/false
Default: true.
During a call transfer, the existing SIP dialog can be terminated by either the transferor or the transferee. When set to true, the KWS will terminate the dialog with a BYE request when acting as a transferor.
Values: true/false
Default: true.
Normally SIP communication on a TLS connection is using the
SIPS: URI scheme.
Disabling this option causes the KWS to use the SIP: URI scheme with a transport=tls parameter for TLS connections.
Values: true/false
Default: true.
By default UDP and TCP transports are disabled when TLS transport is the default. If this setting is true, UDP and TCP are allowed as fallback if TLS fails.
Values: true, false.
Default: false
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Version PCS06A_ - October 13, 2010
This release replaces the PCS05D_ release as the latest generally available (GA) release.
Added or Changed Features
KWS redundancy available.
If a redundancy license is installed it is now possible to configure a redundant setup of
KWS. For more information refer to the application note regarding redundancy.
Major upgrade of the provisioning handling including several enhancements to address the scenarios where a KWS is hosted by a service provider.
Earlier local changes made through the web GUI would not be reverted by provisioning.
This would make provisioning server settings and local settings inconsistent. Now the provisioning server settings, user data and firmware always take precedence over any local changes. I.e:
- The config.xml file on the provisioning server takes precedence over settings on the KWS.
- The users.xml file on the provisioning server takes precedence over user data on the KWS.
- The firmware.bin file on the provisioning server takes precedence over the version on the
KWS. A side effect of this is that provisioning (if activated) will revert firmware if updated via the GUI. I.e. if provisioning is configured and someone upgrades (or downgrades) the firmware from the version that is on the provisioning server, the provisioning process will revert the firmware version to the one available on the provisioning server.
Added provisioning configuration merge functionality. This means amongst other things that provisioning will only initiate a reboot if a setting is changed that requires a reboot to become active. In previous releases any change to the configuration files located on the provisioning server would initiate a reboot.
Eliminated several reboot scenarios. Some settings that previously required a reboot before a change would become active do no longer require a reboot to become active.
Do not require a reboot to enable/disable XML-RPC.
Do not require a reboot to enable/disable MSF.
Do not require a reboot to enable/disable local media-resource.
Do not require a reboot for provisioning method, URL, interval and time.
Do not store default values in config.xml. This means that any setting which is left at the default value will not be saved in the config.xml file. Furthermore empty configuration keys are deleted from the file, thus the config.xml file has become much smaller and radically easier to read.
More robust flash upgrade process. Reboot is now handled centrally to ensure that the
KWS is not rebooted while flashing the firmware.
Add Reboot when idle function. This way it is now possible to schedule a reboot to take place when no calls are active on the system.
Remote syslog improvements.
It is now possible to send debug messages via remote syslog. Furthermore it is possible to configure which log levels to send via remote syslog. All of the changes to the remote syslog can be made without restarting the KWS. This means that e.g. in a hosted environment the provider can increase or decrease the level of logging from a specific KWS
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KIRK Wireless Server 6000, Release Notes without affecting the users of the KWS. In e.g. a trouble-shooting scenario the provider/administrator can increase the log level while debugging and subsequently decrease the level again.
License handling changed to support cumulative licenses. I.e. several licenses can be installed on the KWS independently of each other.
Store licenses in config.xml.
The license was stored outside the configuration but is now moved into config.xml. This makes it possible to handle the license via provisioning.
Distinguish between transport and timeout errors when displaying error messages in the handset.
In earlier firmware versions all transport and timeout errors were reported as being transport errors. Now the transport and timeout errors are reported separately. A transport error is when a request cannot be delivered to the destination due to for example failed DNS lookup. A timeout error is when a response is not received within a reasonable time.
Make SIP client transaction timeout configurable. Increase this time to eliminate timeout errors towards the SIP Provider or decrease it to reduce fail over time if you have several
SIP proxies configured. If you have a “not-so-reliable” connection to your SIP provider/IPBX, it may be an advantage to increase this value. The value specifically controls timer B and F as specified in RFC3261.
Prolonged the time a KWS waits for a response from a DHCP-server during boot before start-up. This addresses an issue where a KWS is configured for DHCP assigned IP-address and the KWS is deployed on certain Cisco switches (e.g. Catalyst 3560). On these switches the network link establishment is delayed which could result in a situation where the SIP
User Agent Server is started before the KWS has received a DHCP assigned IP-address.
Previously the KWS would wait 30 s before starting the KWS without a DHCP-assigned IPaddress, this is now prolonged to 60 s.
Statistics is now handled locally in media resources and base stations. This means e.g. that connection statistics for media resources and base stations will not be reset when a KWS restarts. If a manual statistics reset is performed on the KWS the connection statistics counters will however be cleared.
Log (with level notice) when settings are changed either from the GUI or using provisioning. Indicate whether the change requires a reboot.
Added support for RFC 3326 Reason header. This allows the PBX to control if the handset will display “Missed call” when part of a ring group. As an example if 2 handsets are part of a ring group an INVITE is sent to both handset. If handset 1 answers the call, the PBX can send a CANCEL with reason header “Call Completed Elsewhere” to handset 2, which will result in the fact that no missed call indication will be displayed on handset 2. This is supported by e.g. Asterisk 1.6.
Support for this is expected to become available in the 50xx, 60xx and 70xx series handset
Q1 2011.
Allow "+" in outgoing B-no. If. e.g. "+45123456" is received as a calling party number a redial from the handset will now be allowed.
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Support for this is expected to become available in the 50xx, 60xx and 70xx series handset
Q1 2011.
Add broadcast of 60xx & 70xx series handset XML-RPC messages SMSSetupReq and
ExtendedHwReq.
Handling of SIP registrations improved. A new queuing technique allows for faster SIP registering of endpoints. In connection with e.g. reboots SIP registrations of many users is now handled faster.
If problems are logged in the Media Resource concerning the connection to the KWS, it is now logged which KWS (which KGAP) it is (to handle a redundancy setup with several
KGAPs).
When the media resource logs messages, the message is prepended with MR (serial:####) to distinguish media resource logging from KWS (KGAP) logging. Furthermore shutdown connection to KGAP, and establish connection to KGAP is logged.
More elaborate logging on the KGAP concerning connection establishment towards media resources.
More verbose when ping or traceroute fails.
It is now possible to allow/disallow new media resources or base stations to connect to a
KWS. This can be controlled via the web GUI at Configuration|Wireless Server or through provisioning. Any media resource / base station which is known by the KWS i.e. has been connected before, is allowed to connect regardless of this setting, however new (unknown) media resources/base stations will not be allowed to connect if set to disallow. The default setting is allow.
LogMessage "HL_ME_U_PLANE_ind Me-Instance null. (Pmid:xxxxxx)" downgraded from warning to debug because it is misleading. The warning is logged before an abnormal release but is caused by the abnormal release.
LogMessage "Users download complete" downgraded from info to debug.
LogMessage "Configuration download complete" downgraded from info to debug.
LogMessage "Firmware version download complete" downgraded from info to debug.
Added info LogMessage "Firmware check complete".
LogMessage "Provisioning reboot requested" upgraded from info to notice.
Log a message if someone is trying to access XML-RPC while it is disabled.
Removed Features
Removed debug log level from the message filter on the status logs page in the GUI. The debug messages were not available through the GUI anyway.
The Derived Cipher Key which is used for encryption of voice data in the air is no longer stored at each handset location registration. It is stored at handset subscription and not subsequently. If authentication of calls is enabled a new derived Cipher Key will be calculated at each call. The only reason for storing the Derived Cipher Key after each location registration was to support the scenario where authentication of calls is disabled while encryption of calls is enabled. This scenario is no longer supported, to prevent problems with mismatching Derived Cipher Keys between PP and FP, authentication of calls is no longer optional if encryption is activated. This ensures that a new Derived Cipher
Key is generated at every call.
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Corrections
Handle UPDATE requests correctly for incoming calls. This fixes an issue with some call transfer scenarios encountered on a CUCM and on an Avaya Aura™ Session Manager.
Always set to-tag in SIP responses except 100. This fixes an issue encountered in interoperability testing against Avaya Aura™ Session Manager version 6.0, regarding handling of call forwarding (302 Moved Temporarily).
RTP check that data is coming through the jitter buffer and if not reset the session.
Addresses DECTESC-204 which is an issue with one-way voice in the second of two subsequent calls to the same number on an Aastra PBX.
Do not crash when Referred-By header is missing in REFER, DECTESC-207 Avaya IPOffice.
For outgoing calls only check the ARI and not the RPN, fixes problem with DistyBox 300 reported in DECTESC-208.
When a handset performs an outgoing call the ARI of the system is sent to the system and checked. This failed for DistyBox 300 because the RPN was included in the ARI.
Send BYE from transferor if transfer target does not send BYE.
Addresses DECTESC-215 where a PBX does not send a BYE in an attended transfer scenario causing a hanging call on the PBX. The scenario was identified on an IPECS Call Server.
Bug fixed. May cause configuration daemon to crash if no network.domain is defined.
Disabling MSF did not work, this is now corrected.
MSF was reported as disabled but was still active.
Shutdown uPnP properly if disabled.
Pressing save in the web-GUI for a user in the user list would initiate a new SIP register even in the case where nothing was changed for the user in question. This is corrected.
Indicate “reboot required” for endpoint separate local ports (a setting on the Configuration
| SIP page of the web GUI). This setting requires a reboot and this was not indicated in earlier releases.
Fixed javascript problem in Firefox with auto sync. warning.
Do not truncate the last character of a broadcast request sent via XML-RPC.
Fix bug not allowing CLMS to a single handset via MSF.
Fix group number in CLMS via XML-RPC. Previously it was interpreted as hexadecimal, now it is interpreted as decimal.
Only write DNS configuration when we have something to put in it - avoid overwriting
DHCP.
If HTTP request line contains a full url, strip scheme and host part.
Configuration File Parameter Changes
File Action Parameter
config.xml Added redundancy.failover_time
Description
The time in seconds from a redundancy node detects a failure until it initiates a failover operation.
Default: 15.
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File Action
config.xml Added config.xml Added config.xml Added
Parameter
redundancy.peer redundancy.database_uuid config.xml Added sip.client_transaction_timeout config.xml Added
Release Notes
Q4 2014 redundancy.mode license
Description
The hostname or IP address of the redundancy peer node
Default: none.
Unique ID of the distributed database of the system which must match for replication to be performed.
When reset on the master it is automatically generated, and when reset on the slave, it is retrieved from the master. It must be reset when changing a master node to a slave node or when moving a slave node to another system.
Default: Randomly generated.
Example: 6c71a688-23fc-
4d54-845c-1b80172dd75e
The mode of the node:
Either a normal single node system, a master or a slave node in a redundant system.
Values: "single", "master",
"slave".
Default: “single”
Specifies the timeout for client transactions. This controls timer B and F as specified in RFC3261.
Values: Milliseconds (1000-
32000).
Default: 4000.
Storing of licenses if installed.
Values: A comma separated list of licenses.
Default: Empty
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File Action Parameter
config.xml Added security.allow_new_media_resource config.xml Added security.allow_new_rfp
Description
This setting controls whether new media resources are allowed to connect to the KWS. Any media resource which is known by the KWS i.e. has been connected before, is allowed to connect regardless of this setting, however new (unknown) media resources will not be allowed if this setting is false.
Values: true, false.
Default: true
This setting controls whether new base stations are allowed to connect to the KWS. Any base station which is known by the KWS i.e. has been connected before, is allowed to connect regardless of this setting, however new
(unknown) base stations will not be allowed if this setting is false.
Values: true, false.
Default: true
Version PCS05D_ July 13, 2010
Added or Changed Features
None
Removed Features
None
Corrections
Provisioning: Do not handle SIP NOTIFY check-sync events while updating the firmware.
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If a SIP NOTIFY check-sync event was received while provisioning was updating the firmware, the device could reboot. This could make the device unable to start up afterwards.
This is only a problem if the firmware is updated via provisioning and SIP NOTIFY checksync events are used.
Configuration File Parameter Changes
None
Version PCS05C_ Q3, 2010
Added or Changed Features
Added base station radio synchronization loop detection.
It is of major importance that the configuration of the synchronization of the base stations does not contain loops. To ease the configuration of the system, automatic loop detection is added. The configuration is tested for loops at boot up when a base station configuration is saved and when the Loops button is clicked on the base station administration page.
Be aware that the loop detection might be unable to detect loops correctly if the configuration contains duplicate RPNs or repeaters are involved.
Implemented auto-answer feature which can be used for intercom and loudspeaker call. If an INVITE with an Alert-Info header, a Call-Info header or an Answer-Mode header is received, it is possible to make a Polycom handset automatically answer the call, mute the microphone and turn on speakerphone.
The reason for handling several headers for activating this feature is that different SIP-PBXs have different default implementations. The following list of headers will activate auto answer:
Alert-Info: Auto Answer
Alert-Info: info=alert-autoanswer
Alert-Info: Ring Answer
Alert-Info: info=RingAnswer
Alert-Info: Intercom (This is the default setting on Trixbox)
Alert-Info: info=intercom
Call-Info: =\;answer-after=0
Call-Info: ;answer-after=0
Answer-Mode: Auto (This is according to RFC 5373)
The feature is implemented in the 50-, 60- and 70-Handset series (except 7010 which does not have speakerphone). The feature requires a recent firmware for the handsets.
Handset Firmware version
5020 and 5040 series handsets
6020 and 6040 series handsets
PCS08Ja (or newer) released 2010-Q3
PCS06Da (or newer) released 2010-Q3
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Handset Firmware version
7020 and 7040 series handsets PCS06Da (or newer) released 2010-Q3
Please note that the Alert-Info header can also be used to control external/internal ring pattern. This feature is not affected by the addition of the loudspeaker call feature. The following Alert-Info headers can be used to enable internal ringing (external ringing is default).
Alert-Info: <internal>
Alert-Info: <alert-internal>
Alert-Info: internal
Alert-Info: alert-internal
Central phonebook: Increase the maximum number of records from 10,000 to 40,000. It should be noted that while retrieving phonebook data from a remote LDAP-server, the phonebook will be inaccessible. This means that the refresh interval (the interval at which the central phonebook data is being copied from the LDAP to the KWS) should be chosen with care. The combination of a slow LDAP-server/slow LDAP-server connection and a high number of entries in the corporate phonebook ( > 10,000) should be configured with a long refresh interval, e.g. once-a-day.
Central phonebook: Add support for more dialable numbers per record.
This allows for dialling for example business and mobile phone via the central phonebook.
The configuration parameter phonebook.ldap_number_attributes lists the LDAP attributes containing phone numbers, and the parameter phonebook.csv_number_fields contains
CSV columns containing phone numbers.
Central phonebook: Extended strip number feature to include replace number feature.
This allows for more advanced number manipulation. See the configuration parameter phonebook.ldap_prefixes for details.
Added ping and traceroute to the Web GUI (by popular demand from several customers).
For network diagnostics, a ping and traceroute feature has been added. It is accessible through the Status menu.
SIP UPDATE support (RFC3311).
The KWS is able to receive and handle SIP UPDATE requests.
Added DHCP syslog server support.
A syslog server can be assigned via DHCP option 7.
Added DHCP NTP server support for KWS6000.
A NTP server can be assigned via DHCP option 42.
Added reboot required banner to the Web GUI.
When a configuration parameter requiring a reboot is changed a yellow reboot required banner is shown until next reboot. This is to eliminate situations when administrators have updated configuration settings which require a reboot to become active and subsequently have forgotten to reboot the KWS.
Export of rfps.xml and statistical data as part of the exported log for better diagnostics.
More verbose logging when malformed SIP messages are received.
User’s CSV import is now more verbose on failure. Errors are listed in the log.
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Revised handling of SIP authentication credentials.
The implementation of SIP authentication credentials varies significantly across different platforms and providers. To accommodate this, the SIP authentication credentials handling is made a lot more flexible.
The authentication user is selected with these priorities:
1. Per user authentication user
2. Default authentication username
3. Per user username.
The authentication password is selected with these priorities:
1. Per user authentication password
2. Default authentication password.
All scenarios which were possible before are still possible for example the most restrictive
(and most cumbersome to implement), i.e. using a per user authentication user and per user authentication password. However, it is now also possible to skip entering an authentication user in which case the username will be used as authentication username, and combining this with either a per user authentication password or a default authentication password.
Added tooltips to the SIP configuration and edit user Web pages.
Implemented Connected Line Identification Presentation.
Shows returned display name in handset display for outgoing calls.
If the To header in the 200 OK responses received for an INVITE contains a display name, this is displayed in the handset.
Return of display name to caller for incoming calls.
The display name is added to the 200 OK responses sent for an INVITE.
More pedantic provisioning parsing of config.xml and users.xml to avoid that an incorrect file deletes users or configuration.
Added SW PCS to SIP User-Agent header.
More verbose display of SIP errors in handset.
If a textual error message exists for a SIP error code, this is displayed.
Added support for sending History-Info (RFC4244) used for call forward loop detection.
This can be used to avoid problems when users are making a loop by forwarding calls to each other in a ring. For instance 2000 is forwarding to 2010 which is forwarding to 2020 which is forwarding to 2000.
For incoming calls with no CODEC match, the response with error 488 is earlier.
Sends the error before the handset starts to alert and terminates the call.
XML-RPC: endpoint_base_stations() function added.
This function allows for querying a Polycom handset for its base station map. This can be used for positioning of the handset. See the XML-RPC SDK version 1.4 for further details.
The feature is implemented in the 50-, 60- and 70-Handset series. The feature requires a recent firmware for the handsets.
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Handset Firmware version
5020 and 5040 series handsets
6020 and 6040 series handsets
PCS08Ja (or newer) released 2010-Q3
PCS06Da (or newer) released 2010-Q3
7020 and 7040 series handsets PCS06Da (or newer) released 2010-Q3
Added settings for sending callees preferred CODEC in SDP answers and for only sending a single CODEC in SDP answers. See configuration parameters sip.media.sdp_answer_with_preferred and sip.media.sdp_answer_single.
Removed Features
None
Corrections
Fixed problem with auto creating maximum users.
When auto create users was enabled, it was not possible to auto create the last user of the maximum users.
Fixed hanging call waiting status when handset was released while waiting for a re-INVITE response (on/off hold).
This made the KWS unable to handle call waiting for that particular handset until a reboot.
Fixed wrong dialog handling for call waiting that made the To-tag change between the 180 and the 200 response.
This caused some SIP servers to handle the responses as they were sent from forked calls.
The issue was identified to cause problems on for example Shoretel.
Pressing the alarm button on a subscribed Bosch D6000 handset caused system failure and required a reboot.
Removed RFPI scanner CSV file error from log.
Configuration File Parameter Changes
File Action Parameter
config.xml Added phonebook.csv_number_fields config.xml Added phonebook.ldap_number_attributes
Description
The indexes of columns that contain dialable numbers.
Values: List of indexes of dialable columns.
Default: 2
Example: 2,3
The names of the LDAP attributes that contain dialable numbers.
Values: Dialable attributes provided by the LDAP server.
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File Action
config.xml Changed phonebook.ldap_prefixes config.xml Added sip.media.sdp_answer_with_preferred config.xml Added
Parameter
sip.media.sdp_answer_single
Description
Default: telephoneNumber,mobile
Example: telephoneNumber,mobile
The phone number prefixes to replace or strip, separated by a comma. For example if the phone number is
+45678912345 and the user must dial the 12345 extension, then “+456789” is specified in the strip prefixes field.
If a "=" is added, the prefix will be replaced instead of stripped. For example if the phone number is
+4576280001 and the user must dial the 004576280001 extension, then “+=00” is specified in the strip prefixes field.
Values: Phone number(s) to replace or strip.
Default: "+=00"
Example: "+45,+=00 "
Specifies if the media handling must ignore the remote SDP offer CODEC priorities.
Values: true, false.
True - ignores remote
CODEC priorities.
False - honours remote
CODEC priorities.
Default: false
Comment: Enabling this option violates the RFC3264
SDP offer/answer model.
Specifies if the media handling must provide only a
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File Action Parameter Description
single CODEC in SDP answers.
Values: true, false.
True - provides only a single
CODEC.
False - provides all matching
CODECs.
Default: false
Version PCS05B_ Q2, 2010
Added or Changed Features
Support handling of pauses in phone numbers: This makes is possible to include pauses in dialed phone numbers. If pauses are added in a phone number the part before the first pause is send in an INVITE and the KWS will wait for a 200 OK before sending the pauses and the rest of the number via DTMF. Typical applications for this feature are nurse call system integration or voicemail applications. As an example it is now possible to store the following number in the phonebook/speed dial.
“5555pp8888#” where
- 5555 could be the number to the voice mail application.
- pp would indicate two pauses (this would give the voicemail application time to send out a new dial tone and be ready to receive an access code.
- 8888# would be the access code.
Phone numbers including pauses can be entered on the handset or received as call back numbers via the XML-RPC application interface or MSF application interface (a comma “,” or a p “p” can be used to denote a pause in a call-back number).
Added syslog facility configuration: This makes it possible to configure the source facility used for syslog messages. The default is local0. For further details on remote syslog facilities refer to RFC5424.
Only reset media resource DSP on re-INVITE when needed: If a re-INVITE was received very early in a call it could potentially cause one-way-voice.
The internal messaging feature added in firmware PCS05__ has been improved: Previously, internal messages were echoed on the XML-RPC application interface, this is removed.
Reduced production time: Due to increasing demand and increasing amount of delivered devices the initial creation of an empty file system has been optimized. This only impacts the production process and has no impact on devices in the field.
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Removed Features
None
Corrections
Fixed problem with local call forward in a setup with local call forward enabled and call waiting disabled.
In this setup if a user is in an active call, and a second call is received the system previously would send busy to the second caller. This is now corrected so a second caller will be forwarded.
Fixed not working NTP on media resource.
Mask DECT high priority bit to remove problem with subscribing some Bosch handsets
Removed memory leak when failing to decode SIP replaces header.
Configuration File Parameter Changes
File Action Parameter
config.xml Added log.syslog.facility
Description
Used to specify the remote syslog facility used for log messages. Refer to
RFC5424 for details.
Values: The facility number to be used for the device. An integer between 0 and 23.
Default: 16 (“local 0”)
Version PCS05A_ Jan. 27, 2010
Added or Changed Features
Disable reboot button in KWS base station edit when base station is off line.
Do not generate an error when an off-line base station is edited and saved.
Save XML-RPC SMS on the handset stack when sent using endpoint_sms(). This makes it work the same way as handset-to-handset SMS.
Do not send DECT pages to base stations that are not in sync. They are unable to handle them.
Handle DECT page limiter signals from the base stations.
Send a XML-RPC endpoint_broadcast event as a response to the endpoint_broadcast() function.
Improved MSF buffer handling.
Removed Features
None
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Corrections
Removed fault causing restart when MSF buffer is full.
Limit Message Waiting Indication (MWI) rate to not overload the infrastructure when many
MWI are received.
Removed a NULL pointer dereference when deleting a user and MWI is disabled.
Configuration File Parameter Changes
None
Version PCS05__ Q1, 2010
Added or Changed Features
Call waiting is now supported. It must be enabled to be active (default on). Call waiting is supported on the whole range of Polycom DECT Handsets. However due to differences in keyboard layout, audio processing capabilities and display types, the appearance (audio as well as visual) differs between the different handsets. The solution implemented is a tradeoff between back-ward compatibility and appearance. Note: The 5020 and 5040 handsets require firmware PCS_08Ca or newer.
Accepting a new call: If call waiting is enabled a second call can be accepted by pressing
“R”, in which case the other end will be set on hold and a connection will be established to the new/call waiting caller.
Rejecting a new call: Pressing left arrow/ok button will reject the call waiting call.
Ending the old call and taking the new call: Pressing on-hook while the second call (the call waiting call) is alerting, will terminate the old call and the handset will start/continue ringing. It is now possible to answer the new call.
Ending an established call (if two calls are active):
- If two calls are established due to call transfer pressing on-hook will complete a call transfer.
- If two calls are established due to an incoming Call Waiting which is accepted, pressing on-hook will terminate both calls.
Toggling between two active calls: Pressing “R” will toggle between two active calls.
Ending the active call if two calls are present: Pressing left arrow/ok button will terminate the current call (but not the second call).
Add Message Waiting Indication (MWI) for the 2010 handset. With this addition Message
Waiting Indication is supported on the complete range of Polycom DECT handsets.
Local call forward (unconditional) is now supported. Number to forward to is configurable from the web-GUI as well as directly from the handset. Using the web-gui the Local Call
Forward number can be viewed/edited directly from the user entry of the user in question.
The feature code for enabling/disabling local call forward from the handset can be configured from the “Configuration|Wireless Server” menu. The default code is “*21*$#” where “$” denotes the number to forward to. If a handset has call forward enabled the standby text will be pre-pended with (CFU) to give the user an indication that the handset is forwarded.
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It is now possible to disconnect the active call if two calls are active (either due to an attended call transfer, or due to an accepted call waiting call). If two calls are active pressing left-arrow will disconnect the active call (without disconnecting the in-active call).
Increased string lengths for SIP parameters.
- Default domain 32 -> 256.
- User name 32 -> 64.
- User domain 32 -> 64.
- User authentication 32 -> 64.
Introduced remote syslog (RFC5424) via UDP. The remote syslog allows for using a PC to receive messages/logging from a KWS.
Added internal messaging for sending text messages between handsets without requiring an external application.
The feature is enabled per default but can be disabled if it interferes with an external application.
Failure to read ARI is now logged as EMERGENCY (was KSF_CRITICAL).
MSF/XML-RPC: Release DECT connection immediately when a PP_STATUS_ind initiated by the handset is received.
Support for advanced messaging features introduced. This includes MSF_SMS_SETUP_req
(MSF format 3) and support for MSF_SMS_RESPONSE_ind & ExtenHwReq/Cfm. These features will become available with the release of the upcoming next-generation handset series (the 60xx and 70xx series). The advanced features include alarm buttons, tear-off cord, multicolour LED controllable from an application and motion sensor etc.
Do not send XML-RPC/MSF messages to a handset while messages are queued for the handset.
Added XML-RPC endpoint_release event.
Provisioning improved detection of firmware version inconsistency to avoid problems if firmware is updated manually.
Provisioning is made more verbose. Download of users, firmware and configuration from a provisioning server is now logged to the message log.
Also log line number when failing to parse users.xml.
Do not stop user import if displayname or standby text is invalid or too long - just skip or truncate and log a message.
Do not abort provisioning process when one of the steps fails.
When exporting logs, the message log is stored in clear text. The message log can now be read with standard software.
Improved logging of SIP failures.
Improve log export speed.
Attended transfer: Send the REFER to inactive dialog instead of active. This is required by
Siemens HiPath and Toshiba.
Create a KSF log on media resource crash.
Removed Features
None
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Corrections
Add double-quotes to SIP display names to allow special characters and international letters. This is required by the RFC and e.g. Cisco Call Manager.
Fixed problem with provisioning polling interval. This could result in the fact that the device stopped polling for updates.
Increase SIP dialog local cseq when a request is re-send. This solves a problem with middialog authentication of requests. The problem was originally seen with a Nortel IPBX
(DECT-142).
Does not require that a SIP dialog is established when 180 Ringing is received. Fixes problem with missing dialog parameters for Aastra and Splicecom.
Fixed resolver CNAME problem. DNS CNAME records now supported.
Handles a comma(,) in the username part of the URI in a Refer-To header (DECTESC-167).
Removed a lot of unnecessary writes to the flash. These induced unnecessary tear on the flash, especially during boot.
Changing sip.media.symmetric setting would issue an error in the message log: “Unknown
SIP configuration key: sip.media.symmetric”, this is fixed.
XML-RPC: Fixed problem with zero length data in PP_STATUS_ind.
Removed a few large buffers from the stack. These may have caused sporadic failures.
Minor NTP client improvement which reduces the amount of “NTP failed” errors in the log.
XML-RPC/MSF: Handle PP_STATUS_req/ind in more states.
Report SIP transaction failed if decoding the unauthorized header fails.
Re-classified some log messages.
Removed memory leak when receiving a SIP MESSAGE.
Validate configuration keys when setting them. This avoids malforming the config.xml.
Improved logging of failures in connection with the DNS resolver.
A "No license available” notice message was written to the log even if no license was required, this is corrected.
MSF: Handle XML escaped characters correctly for incoming messages.
XML-RPC/MSF: Clean up release reasons to comply with the documentation.
Normal release reasons (0x00) are unchanged but the values of other release reasons have changed. For XML-RPC refer to the XML-RPC SDK version 1.1 or later for details.
Fixed problem with UPnP UUID not being unique. If more devices on the network have the same UUID, only one of them will be shown when UPnP devices are listed.
Does not require a default gateway for UPnP to work.
Configuration File Parameter Changes
File Action Parameter
config.xml Added log.syslog.host config.xml Added log.syslog.port
Description
Specifies the remote syslog server host address.
Default: Empty
Specifies the remote port of the syslog server.
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File
config.xml
Action
Added
Parameter
feature_codes. enable config.xml Added feature_codes.call_forward.unconditi
onal.enable config.xml Added feature_codes.call_forward.unconditi
onal.disable config.xml
Config.xml
Added
Added application.internal_messaging sip.callwaiting
Description
Values: The port number on a remote syslog server.
Default: Empty which defaults to
514
Used to enable/disable local handling of feature codes.
Values: true/false
Default: false
Specifies the feature code used for enabling unconditional call forward (CFU).
Values: The feature code users must dial to enable unconditional call forward.
Default: *21*$#
Specifies the feature code used for disabling unconditional call forward (CFU).
Values: The feature code users must dial to disable unconditional call forward.
Default: #21#
Used to control if messaging between handsets is handled internally or by an external application. If enabled messages will be handled internally.
Values: true/false
Default: true
Used to control whether Call
Waiting is enabled.
Values: true/false
Default: true
Version PCS04B_ October 20, 2009
Added or Changed Features
None
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Removed Features
None
Corrections
Removed potential media resource problem present in firmware PCS04__ and PCS04A_.
This problem would result in the loss of all active calls on the media resource and a subsequent restart of the media resource.
Removed delay in the media stream after re-configuring media with re-INVITE.
For example, after placing a call on and off hold a delay was introduced in the voice stream.
Configuration File Parameter Changes
None
Version PCS04A_ October 12, 2009
Added or Changed Features
None
Removed Features
None
Corrections
Corrected provisioning check at specific time.
If the device was configured to check for updates at a specific time each day, the device would only check for updates twice.
XML-RPC application interface: The method end_call_display() ignored the setupspec1 parameter.
Removed memory leak related to DECT encryption.
After handling 2,000,000 calls with DECT encryption, the device will run out of memory.
Configuration File Parameter Changes
None
Version PCS04_ Q4/2009
Added or Changed Features
Added support for entering more SIP proxies for failover and load balancing.
This feature is relevant in a setup with more than one SIP proxy. In this case it is now
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KIRK Wireless Server 6000, Release Notes possible to manually enter the SIP URI of the proxies, in earlier releases this could only be done with DNS-SRV.
Added UPnP for discovery of devices.
UPnP is an acronym for Universal Plug and Play. If for some reason, the IP-address of the device is unknown (e.g. forgotten or DHCP-assigned), UPnP can be utilized to easily identify the IP-address of the device. If “My Network Places” in Windows is setup to show icons for networked UPnP devices, every KWS300/6000, Media-resource, and Base station will be present in “My Network Places”.
Added method for manipulating settings by requesting an URL. o http[s]://<host>/config/get?<key> – http://192.168.0.1/config/get?sip.proxy.domain o http[s]://<host>/config/set?<key>=<value> – http://192.168.0.1/config/set?sip.proxy.domain=example.com
Improved jitter buffer.
The sound quality on IP-connections experiencing jitter issues is improved considerably.
Improved the user interface for managing base stations, media resources, clusters and users.
Several improvements are made based upon customer feedback. Previously when e.g. manually editing or adding e.g. users, after pressing "Save" the GUI would present a new screen acknowledging that the user was edited/added ok. On this screen the user had to press "OK". This is now changed so that after pressing save the user is returned to the list.
A dialog screen is only presented to the user if something goes wrong. As a result, the number of mouse-clicks required to do repetitive tasks with regard to editing/creating items in a list has been reduced.
Improved the user interface for central firmware update.
After making a central firmware upgrade of e.g., media-resources and base station, the media-resources/base stations need to be re-booted before the new firmware is active.
The system will continue to run the previous firmware until a reboot of the devices. This allows for a non-intrusive firmware upgrade, which can be done on the system without affecting normal operation. However, this also means that if the devices are not rebooted the system will continue to run on the old firmware. The user interface has been updated to clarify this.
Improved the user interface with respect to the auto-sync feature of base stations.
The auto-sync feature for base stations is only for usage while deploying the system. This was not clear in the user-interface. A more descriptive text has been added and a warning is issued if auto-sync is enabled.
Added XML-RPC application interface.
The new XML-RPC based application interface uses open standards and is easy to use. This interface gives access to the same functionality as the existing MSF interface but is not based on a Microsoft Windows API. The existing MSF interface will not be affected.
Added HTTP/1.1 persistent connections support to the built-in HTTP server. This is mainly done to increase performance on the XML-RPC interface when using HTTPS.
Improved security measures. Formerly every time a dect device would enter the range of the system (making a location registration) the device was authenticated. Starting with this
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KIRK Wireless Server 6000, Release Notes release additional authentication is performed every time a call is established. Furthermore it is now possible to enable dect encryption of voice sent over the air. In previous firmware revisions all dect communication in the air is scrambled, enabling encryption will additionally encrypt voice with an encryption key. A new key will be calculated for each new call.
IMPORTANT NOTICE!! If dect encryption is enabled it is NOT possible to use repeaters on the system.
IMPORTANT NOTICE!! If dect encryption is enabled it requires base station firmware version PCS04__ or higher.
Removed unnecessary warning: HL_ME_RESOURCE_ALLOCATE_req resource already allocated.
Changed the User-Agent name for the provisioning HTTP client.
Removed Features
None
Corrections
Dialog event package – notify dialog terminated when a call is rejected.
Drop RTP packages with unexpected payload without trying to play them.
Do not crash with high load of MSF and message waiting indication (MWI) traffic.
Fixed problem where the maximum CLMS broadcast data length was reduced with one byte.
Do not show 0kB captured when less than 1kB is captured by the packet capture function.
Fixed a bug not allowing the user to enter POSIX time zones via the GUI.
Do not crash when using DNS SRV and deleting a user.
When users are controlled via provisioning – do not indicate users as changed when the handset has reported a firmware version. This caused the system to report the user data as changed when auto provisioning users even with no changes.
Removed crash when attempting to change the standby text for non-KIRK handset.
Handle international characters better in phonebook. The matched part of a search was not displayed correctly when international letters was part of the match.
Make phonebook stop logging a warning when LDAP server is slow.
Configuration File Parameter Changes
File Action Parameter Description
config.xml Added application.enable_rpc Specifies if the XML-RPC application interface is enabled. true – The XML-RPC interface is enabled and applications can connect. false – The XML-RPC interface is disabled.
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File Action Parameter Description
config.xml Added dect.auth_call
Default: false
Specifies if DECT authentication should be used when establishing calls. true – DECT authentication is required when establishing calls. false – DECT authentication of calls is disabled.
Default: true config.xml Added dect.encrypt_voice_data Specifies if DECT encryption should be used for voice calls.
Disabled – DECT encryption is disabled.
Enabled – DECT encryption is enabled.
Enforced – DECT encryption is enforced and calls are terminated if the handset do not support encryption. config.xml Added sip.proxy. domain[2-4] Specifies domain/host name for additional
SIP proxies.
Default: Empty config.xml Added sip.proxy.port[2-4] Specifies port for additional SIP proxies.
Default: Empty config.xml Added sip.proxy.priority sip.proxy.priority[2-4] config.xml Added sip.proxy.weight sip.proxy.weight[2-4]
Specifies the priority for using a SIP proxy.
Proxies with lowest priority will be preferred and higher priorities will be used for failover.
Values: 1-4
Default: 1, 2 , 3, 4
Specifies the weight for using a proxy. If more proxies have the same priority the KWS will do load balancing using the weight to determine how much each proxy will be loaded.
Values: 0-100
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File Action Parameter
config.xml Added upnp.enable config.xml Added upnp.broadcast
Description
Default: 100
Specifies if UPnP support is enabled. If enabled the device will respond to UPnP broadcasts.
Values: true/false
Default: true
Specifies if UPnP announcements are broadcasted. If enabled the device will periodically broadcast announcements.
Values: true/false
Default: false
Version PCS03B_ (Q3/2009)
Added or Changed Features
DECT-97: Add service codes to read system information via handset. Initiated by typing codes and then pressing off hook from the handset. This information can be read from the system.
○ IP address:
***999*00
○ MAC address:
***999*01
○ Server Firmware:
***999*02
Allow custom posix timezone specification strings.
It is now possible to configure the system to show “½-hour time zones”, by entering a posix string
Add revision to User Agent string.
Firmware version can be obtained from traces, inspecting the User Agent field
Include DNS traffic when capturing SIP.
Allow custom capture filters.
Customize the captured data to a trace by entering a filter in pcap format.
DECT-63: New and improved NTP client.
Improved error recovery.
Information for the NTP client included in the log file.
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Add user/password and enable/disable options to MSF.
It is possible to change login username and password for MSF applications (text messaging interface)
MSF functionality can be enabled/disabled
Send unregister and unsubscribe when deleting an endpoint.
Inform the PBX when a DECT handset is deleted.
Clean out parameters in user names received from some PBX’es.
Handle "302 Multiple Choices" - for now just pick the first choice.
Handle SDP in multipart body.
Added timestamp and synchronization statistics duration to rfps.xml.
If SIP registration fails, re-register within a short time and then wait.
Removed Features
None
Corrections
Fixed problem with authentication on some PBX’es.
Fixed problem with wrong answer to SDP update offers.
Fixed timer problem that might break provisioning.
MSF callback number length increased.
Check for required SIP headers before creating a dialog.
Handle timeout for SUBSCRIBE requests.
Retry if SIP subscription fails.
Skip local media resource in central firmware update.
If media resource firmware is updated, the KWS6000 server is not affected even if it acts as local media resource.
Remove require 100rel header from PRACK as this is wrong according to RFC3262.
Improved CODEC card DTMF handling.
DECT-111: Handle MSF timestamps.
Does not crash in some rare call transfer scenarios.
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Configuration File Parameter Changes
File Action Parameter Description
config.xml Added application.enable_msf Specifies if the MSF application interface is enabled. true – The MSF interface is enabled and applications can connect. false – The MSF interface is disabled.
Default: true config.xml Added application.username Specifies the username required for applications to log in.
Default: “GW-DECT/admin” config.xml Added application.password Specifies the encrypted password required for applications to log in.
Default: “f621c2268a8df24955ef4052bfbb80cf”
(password “ip6000” encrypted)
Version PCS03A_ (Q2/2009)
Added or Changed Features
Retrieving a big file from the internal web server no longer blocks the server.
Retain any existing other call when a REFER triggered INVITE fails, otherwise release the handset.
Do not require username in URI in REFER.
Handle "423 Interval to brief" REGISTER response.
Default log level in the GUI increased from INFO to NOTICE.
Add support for international letters using UTF-8.
DECT-83: If no protocol is specified in the provisioning URL then default to TFTP.
DECT-81: Do not repeatedly program flash if version and binary firmware files are inconsistent.
Log an error if configuration XML contains invalid XML.
Add support for keep-alive used by version 18 or later of MSF.DLL.
Send "unknown op" error when an unknown operation is requested via MSF.
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Removed Features
None
Corrections
Fixed bug in Refer-To handling.
Fixed bug in Record-Route handling.
Fixed bug that made the DTMF duration being rounded down to N*80.
Fixed handling of too long dialled numbers.
DECTESC-75: Fixed bug making it impossible to save Wireless Server Configuration.
Disable unsupported media lines correctly.
Parse remote SDP ptime attribute correctly.
Do not send SDP with new version if remote SDP version has not changed.
Only check for remote SDP version changes if remote SDP was received earlier.
Fixed problem with one-way voice when a call is answered during a handover.
Fix bug not allowing MSF multi-byte status requests
– required for RTLS.
Handle MSF call release without call record correctly.
Configuration File Parameter Changes
File Action Parameter Description
config.xml Changed provisioning.server.url Specifies the static boot server URL from where the KWS will retrieve configuration information. The format is
[<protocol>://[<user>:<password>@]]<host
>[/<path>]. Protocol can be either tftp, ftp or http.
It is optional to specify a protocol. If the protocol is not specified the KWS will default to tftp.
Example: ftp://kws:[email protected]/phon es or 192.168.0.1
Default: Empty
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Version PCS03_ (Q1/2009)
Added or Changed Features
Optional individual ports per handsets for SIP signaling. Extend support to SIP PBXs using per port registration.
Cisco Unified Call Manager 6.1 support.
Provisioning: Possible to centralize configuration and maintenance.
Users export to XML and CSV format: Decrease installation and maintenance cost.
Allow adding users with unspecified IPEI: Option of adding handsets without knowing the
IPEI of the handset. Decrease installation and maintenance cost by allowing field subscription of handset(s) and possibility for remote configuration.
Added system wide DECT access code: Possible to create a default DECT access code for all users
– instead of per user (access code in user will overrule the system default value).
Added automatic standby text update. When the standby text is updated (either through the
GUI or through auto-provisioning) the change appears instantly on the handset (no powercycle of the handset is needed).
In overlap dialing send digits when # is pressed. Optional: Default is disabled.
When a user is deleted, unsubscribe the handset: When user is deleted, the handset removes the subscription to the system.
Added RFC3896 Referred-By handling.
Offered rfc2833 payload type (DTMF payload type) can now be configured default is 96.
Add refresh and clear button in base station administration.
CLI / Name display for complete call duration for incoming calls.
Base station lost sync. Ratio / percentage added.
Added BMC/radio configuration.
Removed Features
No longer possible to use local number
– the SIP user name is now used for MSF.
Corrections
Fix base station lost sync. ratio calculation.
Fix DTMF payload type.
Fix order in route sets for SIP dialogs.
Fix statistics for failed MSF calls.
Fix handling of escaped SIP URI parameters.
Pass all parameters and headers from REFER to the sent INVITE.
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Remove http server crash when downloading rfps.xml.
Remove crash on re-INVITE when collecting digits.
Remove crash on INVITE with long From header.
Configuration File Parameter Changes
File Action Parameter
config.xml Added provisioning.server.method config.xml config.xml config.xml config.xml
Added
Added
Added
Added provisioning.server.url provisioning.check.interval provisioning.check.time provisioning.check.check_sync
Description
Specifies how the KWS6000 will obtain the boot server address.
dhcp – obtain from DHCP option 66.
static – use static configured.
disabled – do not check for updates.
Default: dhcp
Specifies the static boot server URL from where the KWS6000 will retrieve configuration information.
The format is
<protocol>://[<user>:<password>@]< host>/<path>. Protocol can be either tftp, ftp or http.
Example: ftp://kws:[email protected]/ phones
Default: Empty
Specifies an interval for checking for updates.
0 – do not check for updates periodically.
>1 – interval in minutes.
Default: 0
Specifies a specific time for checking each day. The format is HH:MM.
00:00 – 23:59
Default: Empty
Specifies how the KWS6000 will react
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File
config.xml config.xml
Action
Added
Added
Parameter
provisioning.users.check provisioning.firmware.kws config.xml Added sip.send_to_current_registrar
Description
to SIP NOTIFY check-sync events.
disabled – do nothing if a check-sync event is received.
reboot – reboot and check for updates.
update – check for updates and reboot if necessary.
Default: disabled
Specifies if the KWS will try to download and import users from the provisioning server.
false – do not check for users.
true – check for users.
Default: false
Specifies the name of the firmware image to use for the KWS6000. The
KWS6000 will check for a version file and a binary file. They must be located as <URL>/<firmware>.ver and
<URL>/<firmware>
Example: kws300-flash.bin
Default: Empty
Specifies how requests outside a dialog are sent if a list of SIP servers is received via DNS SRV.
false – perform a DNS SRV lookup for each request and determine the destination from this.
true – send each request to the server currently holding the registration.
Default: false
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File Action Parameter
config.xml Added sip.separate_endpoint_ports config.xml Added sip.pound_dials_overlap config.xml Added dect.accesscode config.xml Added Sip.dtmf.rtp_payload_type
Description
Specifies if each user should use an individual UDP for its signaling or all users should use the local port defined in the SIP configuration.
false – use one UDP port for all users.
true – use individual UDP ports for each user.
Default: false
Specifies if pressing # while off hook dialing will dial the entered extension.
false – do not dial when # is pressed.
true – dial when # is pressed.
Default: false
Specifies a system wide DECT access code required for subscribing handsets. The access code is from 0 to
8 decimal digits. Access codes assigned for specific users will override this setting.
Example: 1234
Default: Empty
Offered rfc2833 payload type (DTMF payload type) default is 96.
Version PCS02A_ (Q4/2008)
Added or Changed Features
Added cluster handling. This is only relevant for de-centralized installations.
Added support for DECT frequency swap (requires license and base station with firmware
PCS02a_ or later.
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Added phonebook application. This feature offers a centralized phonebook. The formats supported for the phonebook is csv-file and LDAP.
Added enable/disable send date and time to handsets. This feature makes it possible to select whether the date/time should be visible in the handset or not.
Add distinctive alerting by interpreting the Alert-Info SIP header. Use external ring tone as default. If distinctive ring is supported by the IP PBX, different ring tones can be set for the handset to differ between internal and external calls.
Update MWI when a handset subscribes or makes a location registration.
Always respond with 200 OK when a MWI NOTIFY is received. This is done to avoid terminating an existing MWI subscription.
Added automatic MWI retransmission.
Allow for special characters like &_ in SIP authentication user/password.
Allow alphanumeric SIP username.
Implement RFC4235 Dialog state event package. Used for e.g. call pickup support.
Allow for receiving asymmetric RTP (optional, requires media resource with firmware
PCS02A_ or later). This is required to operate with e.g. a Mitel NuPoint voice mail server.
Detect merged invites after a fork and respond with “482 Loop Detected”.
Added full system backup facility. Instead of separate backups of configuration, users etc. everything is now in one backup and it is optional how much is restored.
Standby text length increased from 16 to 24 characters.
Implemented Type-of-Service/DiffServ. Replaced old Quality-of Service approach with new
Type-of-Service approach.
Removed Features
None
Corrections
Corrected error in subscription statistics (subscriptions which failed due to e.g. wrong or missing DECT access code was logged as a success).
Release MSF-call correctly when no CR is assigned.
Fix reversed time zones. GMT time zones were reversed
– GMT+2 meant GMT-2. This has now been fixed.
Version PCS02_
Initial KWS6000 version.
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Outstanding Issues
The following issues will be fixed in a subsequent release
None identified.
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