Thursday, 18 October 2018

Prize Manager 2.0. 2.9.0.20 (BETA)

NEW - All station mode
NEW - Can now group stations and assign prizes to a group of stations
NEW - Station group security
NEW - Now audit cash prizes and accumulators, deletes, restores and creation entries for contests and prizes
NEW - Dispatcher access can now edit winners but not assign
NEW - Can now add barcodes and serial numbers to prizes
NEW - Changed default filter for liners and contests lists
NEW - Ability to directly add a winner without scheduling their prize

FIX - Better improve UI for showing a large number of prizes
FIX - Logout and login
FIX - Show who entered stock
FIX - Draw verify
FIX - Dispatch ref now saved
FIX - Draws no longer allow accumulator contests
FIX - Liner creation allows related contest to be set

Prize Manager 2.0. 2.7.1.5

FIX - Draws are not replicated to repeat schedule entries FIX - Prize draw entries now use name added on PB4 client

Prize Manager 2.0. 2.6.1.11

New - Calendar requests are now more lightweight, better handles busier schedules

Prize Manager 2.0 2.8.0.20 (STABLE)

FIX - Liners on calendar no longer shows deleted liners

NEW - Prize Draw History, pick can be reviewed and verified that it was not tampered with

Wednesday, 17 October 2018

Audio Server v2 2.8.1.27 (STABLE)

Changes since: Audio Server v2 2.8.1.3

NEW - Build with latest Sdp library for s=" " fix

FIX - Multicast recorder inverting caller and studio audio causing incorrect side of call to provide level metering
FIX - Problems with Opus media format lines in reading/writing SDP
FIX - Cpu utilization in minutely log entry needs to be divided by number of cores to be accurate

PhoneBOX 4 Client 4.8.1.46 (STABLE)

Changes since: PhoneBOX 4 Client 4.8.1.28

NEW - Modify "Source Routing - Lines" view
NEW - Codec text logo's  for Systembase / Prima
NEW - Call screener view - Handset answer next icon always showing

FIX - Switching between call log entries can cause the name field to be empty
FIX - On-air preview close button has clickthrough
FIX - Build with latest common / AS2 Opus media format SDP changes
FIX - Build to include codec payload/name lookup in SDP libarary
FIX - Dispo / alert icon showing over the top of line number on active call details header
FIX - Devices Details duplicated intermittently
FIX - Cleanfeed slideout menu button missing on codec until a call has been made
FIX - Conference button not working
FIX - Add logo for Nica X codec
FIX - RTE - null reference with call screener view on dialing
FIX - Occasional non-switch to outgoing calls in popped call details

PhoneBOX (General) 3.8.1.47 (STABLE)

Changes since: PhoneBOX (General) 3.8.1.32

NEW - Finish Systembase codec implementation

FIX - LW recorder not triggering correctly from PhoneBOX
FIX - Alter TransferImmediate operation to remove call from existing device before moving to new device
FIX - Handset / VX call recording not working
FIX - No media number when answering call to AsV2 device
FIX - Do not use 0.0.0.0 and temporary port when re-inviting handsets with known a provider endpoint
FIX - Prevent handset invites containing codecs not supported by provider during phonecall
FIX - Non preferred G711 is removed from SDP offers
FIX - 3.8 SIP server installer overwriting loglevels.config on each update
FIX - Missing fmtp attribute on SDP prevents OPUS calls working
FIX - Rebuild to include SDP library fix for payload lookup from codec name
FIX - Error when dialling out on AS2 device
FIX - Calls answered call to handset device get stuck in "actively answering" state preventing unpark

OASIS 1.8.1.26 (STABLE)

No changes since 1.8.1.19


SkypeTX 1.8.1.23 (STABLE)

No changes since 1.8.1.13.

PhoneBOX (General) 3.9.0.92 (BETA)

Changes since: PhoneBOX (General) 3.9.0.82

NEW - Improve Aeta codec, add CodecType.Tieline
NEW - Improve connectivity of Element switcher type
NEW - Critical system event emails

FIX - Element switcher type not querying input states on startup
FIX - Alter TransferImmediate operation to remove call from existing device before moving to new device
FIX - LW recorder not triggering correctly from PhoneBOX
FIX - Call log search fails on caller name
FIX - E164 only applied to numbers longer than minAreaCodeLookup

PhoneBOX 4 Client 4.9.0.75 (BETA)

Changes since: PhoneBOX 4 Client 4.9.0.66

NEW - Improve Aeta codec
NEW - Enhancements to Source Routing - Lines view

FIX - Add missing translations on codec slideouts and other areas
FIX - Cleanfeed button not opening dialog
FIX - Memory leak fixes:  make component controls of call log items not derived from view model base
FIX - Translations not showing in view picker unless ini file locale override is present
FIX - Smart queue buttons not appearing properly in location and identity modes
FIX - Buttons at foot of "New status" screen truncated
FIX - Name textbox missing from  call log search
FIX - Short tweet exception when checking for retweets

Audio Server v2 2.9.0.60 (BETA)

Changes since: Audio Server v2 2.9.0.54

NEW - Add HTTP Proxy configuration support
NEW - Refactor Anywhere call and relay device to allow cloud relay server extension

FIX - Multicast recorder inverting caller and studio audio causing incorrect side of call to provide level metering
FIX - LW Recording can fail intermittently
FIX - Error during multicast MOH receive can stop MOH from working

OASIS 1.9.0.47 (BETA)

Changes since: OASIS 1.9.0.43

NEW - Store outbound SMS (incl replies) in the Outbound table
NEW - Include SMS replies in message searching

FIX - Restart reimports custom messages from SQL table
FIX - Import sql datatable blank externalID, relatedexternalid matching to blank externalid
FIX - Typo fix OnAPISMSReceivedArgs renamed to ApiSmsReceivedEventArgs

SkypeTX 1.9.0.57 (BETA)

Changes since: SkypeTX 1.9.0.42

FIX - Add .Net 4.7.1 to installer as prerequisite

Tuesday, 9 October 2018

Audio Server v2 2.9.0.54 (BETA)

Changes since: Audio Server v2 2.9.0.23

FIX - Relay call - audio to handset from browser is incomplete
FIX - WDM channels stereo settings not working properly

OASIS 1.9.0.43 (BETA)

Changes since: OASIS 1.9.0.26

NEW - Rest API binds to http://*, no longer needs INGEST_RESTAPI_HOST or RESTAPI_HOST settings

FIX - Handle errors with calculating message dictionary size & no show group on analytic provider
FIX - Tighten up purge, change index

PhoneBOX (General) 3.9.0.82 (BETA)

Changes since: PhoneBOX (General) 3.9.0.50

NEW - Implement active directory integration
NEW - Improve Aeta codec, add CodecType.Tieline
NEW - Add .Net 4.7.1 to installer as pre-requisite

FIX - Encryption aspect of install process causes errors during setup and web manager fails after install
FIX - Anywhere DLLs missing from Eagle SIP installer
FIX - Anywhere not connecting to live production cloud server
FIX - No media number when answering call to AsV2 device
FIX - Do not use 0.0.0.0 and temporary port when re-inviting handsets with known a provider endpoint
FIX - Non preferred G711 is removed from SDP offers
FIX - Prevent handset invites containing codecs not supported by provider during phonecall
FIX - PB3 SIP - Direct dial from Linksys/SPA phone fails
FIX - Winners list person.Name if available
FIX - Inspect, compare and pass 'ptime' values in handset reinvites
FIX - Ensure SDP origin is consistent between SDP changes
FIX - Prevent codecs being downgraded with provider SDP after intial codec upgrade
FIX - Double accept (OK) messages being sent on incoming calls answered to handset devices
FIX - Stuck call on handset device after service not available dial failure
FIX - INVITE from SIP provider without SDP not answering correctly
FIX - Web manager not using proxy for anywhere requests

PhoneBOX 4 Client 4.9.0.66 (BETA)

Changes since: PhoneBOX 4 Client 4.9.0.46

NEW - French translation tweaks
NEW - Add .Net 4.7.1 to installer

FIX - Right hand menu bar not reset when unpopping a message
FIX - Tick and cross buttons truncated on SMS reply popup
FIX - Bionic Studio logo and show name not aligned
FIX - HD indicator not showing
FIX - Respond button should not be enabled for SMS with no 'reply' account configured
FIX - Client FATAL when selecting 'Source Routing - Lines' view
FIX - Reading panel not updating with call details in 'Presenter Simple' view
FIX - Route in 'Presenter Device' view closes OAQ
FIX - On-air preview close button has clickthrough
FIX - Services dropdown in telco dial pad showing erroneous entries when Anywhere or Skype services exist
FIX - Ensure the client updater/install will downgrade as well as upgrade
FIX - Connection Wizard checking for active directory groups when AD is not enabled
FIX - Search boxes, call info renamed to exact


Monday, 17 September 2018

Audio Server v2 2.9.0.23

Changes since: Audio Server v2 2.7.0.7

NEW - Add WebRTC components
NEW - Add support for SDP fmtp attribute with OPUS
NEW - Build with latest Sdp library for s=" " fix
NEW - Add TURN support from audio server for WebRTC calls
NEW - Add Relay calls for WebRTC
NEW - Add anywhere endpoint configuration from ini file
NEW - Implement support for more OPUS  bitrates
NEW - Add ability for OPUS bitrate to follow what is being sent
NEW - Refactor to take phonebox reference away and move SDP class library to BBCommon.Sip
NEW - Build with latest Sdp library for s=" " fix

FIX - Problems with Opus media format lines in reading/writing SDP
FIX - Anywhere common files not installing properly resulting in Anywhere failures
FIX - Cpu utilization in minutely log entry needs to be divided by number of cores to be accurate
FIX - Sdp differences cause double speed audio on WebRTC call when Firefox makes remote call
FIX - Delay on relay device calls
FIX - Fix reference to BBCommon in Audio.v2 library project
FIX - Slight audio delay issue persisting for WebRTC calls
FIX - WebRtc.dll is not versioned correctly
FIX - Delay building up over time for webrtc calls
FIX - Anywhere segfault caused by answer being set twice on call park
FIX - Prevent Anywhere web socket disconnection
FIX - improve call recording to network paths
FIX - benign error during purging when default ringtone wav is present
FIX - Problems with Opus media format lines in reading/writing SDP
FIX - Cpu utilization in minutely log entry needs to be divided by number of cores to be accurate
FIX - Opus sdp interpretation of channels segment of 'a' line should not affect stereo/mono behaviour
FIX - Build to incorporate positive custom payload number fix for skype and vx handsets

PhoneBOX 4 Client 4.9.0.46

Changes since: PhoneBOX 4 Client 4.7.1.48

NEW - implement prize accumulators in contests
NEW - Implement Active Directory functionality in client
NEW - Add alterations to On Air - Presenter views based on High Visibility flag
NEW - liner read improvements, prevent reread within 5mins
NEW - Implement Active Directory functionality in client
NEW - badge on contents button to show unpopped contests or liners, styling improvements
NEW - Enable multi-lingual UI
NEW - dual name field
NEW - Luci codec implementation
NEW - add support for SMS send account type
NEW - Populate point with message when calling back from an SMS
NEW - Implement message pausing
NEW - Modify "Source Routing - Lines" view
NEW - Upgrade to .Net 4.7.1 to accomodate latest Skype TX component
NEW - French translations  -
NEW - Add BC50 Camera preview and zoom support for Virtual Director
NEW - Upgrade to .Net 4.7.1 to accomodate latest Skype TX component
NEW - French translations  -
NEW - allow simple cash prize draws
NEW - Create translation keys for skins, call states and message routing
NEW - Create translation keys for skins, call states and message routing
NEW - Build with updated SIP stack
NEW - Build with SIP stack containing datagram fragmentation and PRACK auth
NEW - Add ini file setting to change application language (resources culture)
NEW - PB Client Updater service requiring .net 2.0 framework to be separately installed, change to 4.x to match client itself
NEW - Add support for chat within created anywhere invites
NEW - Add support for custom message type
NEW - implement reply button for SMS
NEW - Add anywhere invite chat fields and add show/client machine to create event
NEW - Allow history to be displayed for person or number
NEW - Opt-in mode for call details
NEW - allow copy of SMS phone number
NEW - prizeitem.competitionguid no longer needed
NEW - Add feedback dialog after sending anywhere invite
NEW - Support PM2 station group changes
NEW - Add option to exclude warning / banned callers from Answer Next
NEW - Make handset selection indication consistent with other devices
NEW - Add anywhere formatted email including station name
NEW - Add option to exclude warning / banned callers from Answer Next
NEW - Style web RTC / anywhere invite pad
NEW - Add Bionic Studio logo to application
NEW - Add annotate button
NEW - Style web RTC / anywhere invite pad
NEW - Add promote to video option to device and indication on line
NEW - Enable receiving protoBuf-serialized messages
NEW - Display significant winner alert
NEW - Add gender indication to Device Only views
NEW - Add hotkey for "New Message"
NEW - Add ability to sort call log by gender
NEW - Add option to disable auto client update
NEW - Add ability to see prize winning history directly from a message
NEW - Add "Yesterday" option to call search
NEW - Add ability to drag call from TBU to the on air queue
NEW - Indicator on line when call is 'HD'
NEW - Make copyright date dynamic
NEW - Add Selfop view without clock and chat
NEW - Sdp rework changes to client
NEW - PrizeDrawEntryArg to contain prize name and contest name
NEW - Increase sending of twitter messages char limit to 280
NEW - Codec text logo's  for Systembase / Prima
NEW - call screener view - Handset answer next icon always showing
NEW - Build with SIP stack containing datagram fragmentation and PRACK auth
NEW - Add option to exclude warning / banned callers from Answer Next
NEW - Enable receiving protoBuf-serialized messages
NEW - Allow history to be displayed for person or number
NEW - Add Media button and split Director button functions
NEW - Add Post To File account  type to client.
NEW - Add close button to anywhere invite dialpad
NEW - log entry when the selected device is changed by the user
NEW - Build to incorporate new common components

FIX - Build with latest common / AS2 Opus media format SDP changes
FIX - Build to include codec payload/name lookup in SDP libarary
FIX - PB4 - Call screener - Reject button symbol is missing
FIX - Dispo / alert icon showing over the top of line number on active call details header
FIX - Devices Details duplicated intermittently
FIX - Cleanfeed slideout menu button missing on codec until a call has been made
FIX - CTD in call screener view during call switch
FIX - Cursor jumping in the address field (Screener View)
FIX - Memory leak fixes:  make component controls of call log items not derived from view model base
FIX - Contest winner - prize selection UI improvements
FIX - scheduled cash prize with multiple prizes
FIX - Occasional non-switch to outgoing calls in popped call details
FIX - Conference button not taking effect
FIX - Hanging up conference using lines control dialog causes system hang
FIX - Hangup slideout button not working on devices or destinations for telco calls
FIX - Updating inactive call log point loses last few characters
FIX - Person search issue in call screener handset and 3 column call details views
FIX - Complete UI translation .resx files
FIX - contest and liners text is clipped
FIX - Draw picker does not update existing call records
FIX - Cash accumulator bug leftover funds
FIX - Routing docked news view not working
FIX - picking winners with date override confuses winner info
FIX - refreshing contest liners no longer hangs UI thread
FIX - FATAL - onAirViewPresenter with contest in onair queue
FIX - Complete UI translation .resx files
FIX - codec slideout button bar not working
FIX - Log entry error with view model cache manager
FIX - Device slideout button click through causing call transfer to other TBU
FIX - Memory leak on switch converter case on call log gender
FIX - dial from SMS call back forces G711U, should use auto codecs
FIX - Presenter - On Air Simple and Presenter On-Air Devices view selection is reversed
FIX - click through from message viewer opens pop out chat
FIX - Hangup slideout button not working on devices or destinations for telco calls
FIX - CTD when closing Virtual Director Media pane when Virtual Director is not set to the same show as PB4.
FIX - Complete UI translation .resx files
FIX - CTD when closing Virtual Director Media pane when Virtual Director is not set to the same show as PB4.
FIX - Complete UI translation .resx files
FIX - routing docked newsroom view fatals
FIX - Clicking on HD indicator causes park/unpark
FIX - Client garbage collection happening infrequently enough to raise concerns
FIX - Rebuild with new BBCommon after VX softphone issue
FIX - BBCommon error when loading in a softphone configured device layout with PB VX
FIX - Cursor jumping to start of field when entering details
FIX - Leaving cursor in Name or Location field prevents details switching
FIX - Vx softphone audio only coming out of 1-ear
FIX - Complete UI translation .resx files
FIX - Softphone audio only coming out of 1 ear
FIX - No softphone audio
FIX - calling a skype call from the call log passes the wrong value
FIX - Virtual Director Transcription search fixes.
FIX - IPO installer no longer working
FIX - call search popup should close on clicking 'call' button
FIX - If SkypeTx Codec cannot login to begin with (at startup) you cannot log into another account
FIX - Skype logo disappears from codec in routing view during drag operation
FIX - call log item point text not updated when closing popped entry
FIX - call details sometimes doesnt switch to new active call when calling back
FIX - Virtual Director (Aardvark) doesnt automatically reconnect.
FIX - Virtual Director Media Stills appear blank when connected to Virtual Director Server.
FIX - Contest tab- slow popping of multiple prizes without images
FIX - Cursor jumping to start of field when entering details
FIX - Force Device configuration doesn't get forced as expected
FIX - Stop skype devices appearing in answer control popup
FIX - Certain dial situations not specifying a codec enum and resulting in payload 0 erroneously (G711u)
FIX - Inactive calls added to OAQ are missing details after a client restart
FIX - Size Skype video preview frame properly on client
FIX - Debounce rapid clicks on call buttons
FIX - liner read count not incrementing from 0 until refresh
FIX - call log person entry goes black once call is in progress
FIX - Message pause timeout 15 second debug mode is in release build
FIX - Unable to tick checkboxes in Windows Classic Theme
FIX - Calls that are set to be unblurred become unblurred due to other call activity
FIX - Instagram profile showing 0 followers
FIX - Don't park calls on right click, only left
FIX - VX voip softphone not releasing port after call drops
FIX - prize accumulator bug contest rework
FIX - draw picker issues, including FATAL when no prizes
FIX - Incoming chat text currently can't be selected for copy & paste
FIX - Build with latest common / AS2 Opus media format SDP changes
FIX - Build to include codec payload/name lookup in SDP libarary
FIX - Dispo / alert icon showing over the top of line number on active call details header
FIX - Devices Details duplicated intermittently
FIX - Cleanfeed slideout menu button missing on codec until a call has been made
FIX - Conference button not working
FIX - Add logo for Nica X codec
FIX - RTE - null reference with call screener view on dialing
FIX - Occasional non-switch to outgoing calls in popped call details
FIX - Presenter - On Air Simple and Presenter On-Air Devices view selection is reversed
FIX - codec slideout button bar not working
FIX - Log entry error with view model cache manager
FIX - Device slideout button click through causing call transfer to other TBU
FIX - Memory leak on switch converter case on call log gender
FIX - dial from SMS call back forces G711U, should use auto codecs
FIX - Hangup slideout button not working on devices or destinations for telco calls
FIX - Vx softphone audio only coming out of 1-ear
FIX - Rebuild with new BBCommon after VX softphone issue
FIX - Clicking on HD indicator causes park/unpark
FIX - Softphone audio only coming out of 1 ear
FIX - calling a skype call from the call log passes the wrong value
FIX - Smart queue messages not displayed in client
FIX - Log entry shows DeviceModelSkype for telco devices
FIX - Dragging an empty line to device can change the active device selection without updating the indication
FIX - Docked Devices - Call log item goes black
FIX - Prevent dialling on occupied device
FIX - Skype icon appears on popped call log for active codec call
FIX - Memory leak with added unpopped call log items with call log in view
FIX - Unpopped codec call log entries not showing correct icon
FIX - Call details not switching on new call under certain customer conditions
FIX - tag not recalled from previous calls in the lookup window
FIX - Missing BBCommon network in PB client project
FIX - Virtual Director Media Stills appear blank when connected to Virtual Director Server.
FIX - Virtual Director (Aardvark) doesnt automatically reconnect.
FIX - two devices can be active in 'General / 3 column - call details' view
FIX - Virtual Director Video Stills are not displaying when connected to VD2
FIX - Cursor jumping to start of field when entering details
FIX - Force Device configuration doesn't get forced as expected
FIX - Stop skype devices appearing in answer control popup
FIX - Certain dial situations not specifying a codec enum and resulting in payload 0 erroneously (G711u)
FIX - internal note message types not returned in message search results
FIX - Issues with manual record - Softphone + ASv2
FIX - 'display point' updates are replicated to the wrong field on other clients
FIX - Message search results counter not update
FIX - Social only view doesn't allow Virtual Director show selection
FIX - Message log search - tag filter not applied to results
FIX - call log search results unreliable when 'latest only' applied to call log
FIX - Cursor jumping to start of field when entering details
FIX - Call switching when making outbound calls sometimes makes a black call details header
FIX - Build for opus fixes in Sdp and As libraries
FIX - Skype avatars missing
FIX - issue with multiple skype services in a single line layout
FIX - Ringing skype lines on tabbed page has no indication on the tab
FIX - Crash but using call button from OAQ
FIX - Cursor jumping to start of field when entering details
FIX - If Virtual Director disconnects subsequent videos are not populated in the media tab once Virtual Director reconnects unless the client is restarted.
FIX - Adding message tag whilst paused fails
FIX - A message filter applied when paused doesn't work correctly
FIX - Build to incorporate positive custom payload number fix for skype and vx handsets
FIX - Fatal when using VD carousel
FIX - Prevent second calls being made on VOIP handsets in Avaya mode
FIX - IPoffice prevent being able to add two calls to handset
FIX - unable to reply to tweets within the location smart queue
FIX - Screened SkypeTx lines clickthrough
FIX - SkypeTx if marked as screened Pink stays on line after Skype call has been dropped

OASIS 1.9.0.26

Changes since: OASIS 1.7.1.25

NEW - Remove plain text passwords
NEW - Add mechanism for SMS "reply" account
NEW - Implement SMS Send account type
NEW - Change social media message comms to use more efficient serialization and less bandwidth
NEW - Add PostToFile Social Medai type for Virtual Director Sharing
NEW - Alexa improvements - forgotten checkin
NEW - Add support for custom message type
NEW - Implement SMS send code for ARNStudio
NEW - Add support for custom message type
NEW - Implement SMS send - infrastructure
NEW - implement send SMS code for Fonix SMS provider
NEW - Implement send code for Modica SMS provider
NEW - Add support for custom message type
NEW - Implement SMS send - infrastructure
NEW - Implement internal rest API authorisation
NEW - Add key to 'Voice' ingest
NEW - Set default purge setting to be 180 days with an explicit system setting added to the table for visibility in OASIS admin. Settings are MAX_MESSAGETABLEMETHOD=AGE|SIZE & MAX_MESSAGETABLESIZE=180  (days)
NEW - Use access token for alexa comms
NEW - Build with BBCommon fixes for Rest Api self-hosting
NEW - Add social media activity and sentiment publish
NEW - Remove plain text passwords (DUPLICATE)
NEW - Always overwrite config files to ensure binding redirects are deployed
NEW - Use targets file to import self-hosting (Owin) files properly

FIX - Oasis Exporter sends playout messages to every virtual director back end type according to whether it has a publish queue regardless of whether the playout show guid matches the publish queue guid.
FIX - Remove 00 being automatically appended to SQL table ingested SMS numbers
FIX - description in oasis admin for custom message type Sql fields is incorrect
FIX - custom message sql not correctly reading values
FIX - set user agent on licence web requests to identify our application
FIX - Remove 00 being automatically appended to SQL table ingested SMS numbers
FIX - Error following obscenity filtered keyword in Facebook post stops following posts from being imported
FIX - server CPU alerts can misreport actual CPU usage
FIX - Server doesn't pick up publish queues added via API until restart
FIX - Dynamic column names in SQL SMS ingest not working
FIX - Draws multiple entries for same message
FIX - Message search SP not working
FIX - Normal analytics pages not updating (followed/mentioned/retweeted)
FIX - Re-write client connection and server-side comms for improved scalability
FIX - internal note message types not returned in message search results
FIX - twitter lists for followee analysis not always created for some shows
FIX - Assembly manifest definition mismatch during Rest API startup
FIX - improve logging when a facebook wall reader stops due to token issue
FIX - Creation of person record by SMS receiver should place NULL in the name field rather than the number
FIX - Validation error message now shown for account saving
FIX - Erroneous use of Account Groups in Account object
FIX - Error following obscenity filtered keyword in Facebook post stops following posts from being imported
FIX - server CPU alerts can misreport actual CPU usage
FIX - Switching smart mode should refresh results
FIX - ensure characters such as &, used in posts, are correctly escaped when starting live streaming
FIX - Message search SP not working
FIX - Normal analytics pages not updating (followed/mentioned/retweeted)
FIX - internal note message types not returned in message search results
FIX - Adding more than 1 SMS account to a draw results in double entries
FIX - Instagram logging too much detail at Info verbosity

SkypeTX 1.9.0.42

Changes since: SkypeTX 1.7.1.7

NEW - Add video support
NEW - Integrate new version of Skype TX automation and target .Net 4.7.1
NEW - Implement reliable avatar updates
NEW - Add internal security call header to phonebox start handset call request

FIX - Skype TX softphone audio not working due to payload/SDP discrepancy
FIX - Installer failure
FIX - Installer error when installing service
FIX - Set user agent on licence web requests to identify our application
FIX - Raise channel state change event after channels cleared back to 'ready'
FIX - Skype tokens refreshed automatically not being updated in the database
FIX - Change installer to use localized service login for NT AUTHORITY\NETWORK SERVICE
FIX - Change registry creation code to use network service localized
FIX - Ensure startup of STXC clears SinkAudioVideoIds and SourceAudioVideoIds registry keys
FIX - Remote device (handset) audio not working for a new call
FIX - Errors starting up due to missing DefaultDeviceGuids registry subkey in channel directory
FIX - Ensure vidi and vido files are created on machines without any video device installed
FIX - Raise channel state change event after channels cleared back to 'ready'
FIX - Skype tokens refreshed automatically not being updated in the database
FIX - Change registry creation code to use network service localized
FIX - Remote device (handset) audio not working for a new call
FIX - Raise channel state change event after channels cleared back to 'ready'
FIX - Rest API fix
FIX - Fix Rest API self-hosting references
FIX - Remote device (handset) audio not working for a new call
FIX - Change registry creation code to use network service localized
FIX - Raise channel state change event after channels cleared back to 'ready'
FIX - Fix to common library for answering flag on calls
FIX - Audio delay starting
FIX - Rest API not starting correctly in service mode
FIX - Build to incorporate positive custom payload number fix for skype and vx handsets
FIX - Internal call message handler not registered correctly in SkypeTX.Service Rest Api start up

PhoneBOX (General) 3.9.0.50

Changes since: PhoneBOX (General) 3.7.1.43

NEW - Add Anywhere / Web RTC device, handset and chat support
NEW - Implement Active Directory integration
NEW - Remove plain text passwords
NEW - SIP enhancements to PRACK, UPDATES, REINVITE and registration interval adjustment
NEW - Inbound CLI manipulation - more advanced rules  & E164 support
NEW - Opt-in mode for call details
NEW - inbound CLI manipulation - more advanced rules  & E164 support
NEW - Implement skype video promotion for services
NEW - Provide better auditing data when looking at draws
NEW - Complete rewrite of SIP audio device handset class
NEW - Systembase codec implementation
NEW - Allow simple cash prize draws
NEW - Dual name field
NEW - Winner alerts change
NEW - Add support for Luci Studio codec
NEW - Support PM2 station group changes
NEW - Implement API key approach to allow us to determine whether incoming calls are external and authorised or internal
NEW - Winner alerts change
NEW - Grace mode licencing following a hardware change
NEW - Prize draw entry to contain how it was selected
NEW - Add show setting to control default message log pause timeout in client
NEW - Add service registration / availability status to the API
NEW - Prize accumulator
NEW - Implement "sticky point"
NEW - Upgrade to .Net 4.7.1 to accomodate new Skype TX component
NEW - Liner - prevent re-reads within 5mins
NEW - Build to include datagram fragmentation of SIP over UDP
NEW - SQL part of installer needs to be able to use TLS 1.2
NEW - Add phonebox signal server configuration management and authentication
NEW - Add CORS support to rest API startup to fix OAQ HTML
NEW - Changes to support using message text as point on SMS callbacks
NEW - Add ability to see prize winning history directly from a message
NEW - Add HD indicator to phonecall object
NEW - Change default font size for chat text
NEW - Add new self op simple view
NEW - Provide a separate config file for log levels
NEW - Add master machine studio config field which determines which studios are recorded based off currently connected clients
NEW - PrizeDrawEntryArg to contain prize name and contest name
NEW - Remove v3 client from installer
NEW - Add ability for a machine to be a master machine, signalling which studio/shows are currently in use
NEW - Allow Skype service to use allocated MOH source
NEW - Configurable system option 'hide sensitive data'
NEW - prevent router reconcillation from ever removing inputs and outputs automatically
NEW - SQL part of installer needs to be able to use TLS 1.2
NEW - Remove pm1 pages from webmanager
NEW - Add REST API call to get system guid
NEW - Add studio controller for fetching studio configuration
NEW - Add additional logging for call add and removal

FIX - Missing fmtp attribute on SDP prevents OPUS calls working
FIX - Early media ReInvites that send Update causing dial failure
FIX - Parking out of conference on VX devices not working
FIX - Direct dialing from handsets  not working
FIX - Occasional no audio after transfer immediate between handset devices
FIX - newer config sections missing from menu in webmanager
FIX - Error when dialling out on AS2 device
FIX - cannot create new system
FIX - additional available cash available not calculated
FIX - Cash accumulator bug leftover funds
FIX - Wrong binding redirect for Microsoft.Owin
FIX - Build with SIP stack change for early vs established race condition
FIX - set user agent on licence web requests to identify our application
FIX - 3.8 SIP server installer overwriting loglevels.config on each update
FIX - VD Connection not binding to all NICS
FIX - Server crash in audio device handset
FIX - Skype devices can be deleted when referenced by device layouts leaving orphan records
FIX - Installer repair removes key settings included encrypted password
FIX - Dolphin installer issues with client / as2 updaters
FIX - Installer issues
FIX - station manager can remove users from roles in other stations!
FIX - Service installers not working on first time Eagle install
FIX - Skype devices can be deleted when referenced by device layouts leaving orphan records
FIX - Anywhere session ID not send when placing on hold
FIX - Installshield - remove 'googlephonenumber' from list of optional components
FIX - Cannot save service configuration when anywhere service not selected in webmanager
FIX - Anywhere session ID not send when placing on hold
FIX - Skype tokens refreshed automatically not being updated in the database
FIX - If SkypeTx Codec cannot login to begin with (at startup) you cannot log into another account
FIX - Search on number 2 not working
FIX - Ensure all channels are set to "no device" on startup to clear last device settings on STXC channels after crash / restart during handset call
FIX - v3 client stops displaying other routes on devices after a route change
FIX - PM2 conectionstring.config removed during update
FIX - call log search for names with apostrophe fails
FIX - Ringing handset problem with late provider SDP.
FIX - Lookup code refactor
FIX - Every other email fails when using SSL
FIX - Fix other person records getting picked up while blank anywhere name entered
FIX - SkypeTx Codec - inaccessible ringing call
FIX - Skype TX codec calls cannot be answered in some instances
FIX - Not finding image while creating anywhere email as a service
FIX - Fix to add 'SettingUpCall' state to Skype possible states and avoid removing calls from lines during answering phase
FIX - System.Web.Http.Cors.dll missing from server install
FIX - ensure Skype TX reads the most recent MSA token from the database at login
FIX - ensure Skype TX reads the most recent MSA token from the database at login
FIX - prevent asymmetrical calls with different media types
FIX - Phonebox device API not returning point
FIX - Make sure to Increment Sdp version correctly
FIX - Rest Api errors in server log file
FIX - Fix references for Rest Api hosting
FIX - Missing scripts causing various issues
FIX - Fix references for Rest Api hosting
FIX - Add Skype To Device layout - Back button "File not found"
FIX - draw picker qualifiers - fixes fatal when no prizes
FIX - Missing fmtp attribute on SDP prevents OPUS calls working
FIX - Rebuild to include SDP library fix for payload lookup from codec name
FIX - Error when dialling out on AS2 device
FIX - Calls answered call to handset device get stuck in "actively answering" state preventing unpark
FIX - Early media ReInvites that send Update causing dial failure
FIX - Parking out of conference on VX devices not working
FIX - Direct dialing from handsets  not working
FIX - Occasional no audio after transfer immediate between handset devices
FIX - Build with SIP stack change for early vs established race condition
FIX - 3.8 SIP server installer overwriting loglevels.config on each update
FIX - VD Connection not binding to all NICS
FIX - Build to include SIP stack 1.8.1.6
FIX - Wrong payload type preventing VX softphone from working
FIX - Incoming calls answered to handsets are overriding provider codec list with generic list
FIX - Dolphin installer issues with client / as2 updaters
FIX - Skype codec can get stuck in a ringing out state if dial fails
FIX - dial using handset buttons always requires the provider to use G711U
FIX - Calls left ringing after ending call made from a Cisco SPA handset
FIX - SDP comparison failing when optional number of channel parameter included in media description
FIX - HD indicator on SIP phonecalls not reliable
FIX - Dialling from client with specific codec should offer that and lesser codecs in the INVITE
FIX - dial using handset buttons always requires the provider to use G711U
FIX - HD indicator on SIP phonecalls not reliable
FIX - Ensure all channels are set to "no device" on startup to clear last device settings on STXC channels after crash / restart during handset call
FIX - v3 client stops displaying other routes on devices after a route change
FIX - Skype tokens refreshed automatically not being updated in the database
FIX - If SkypeTx Codec cannot login to begin with (at startup) you cannot log into another account
FIX - Search on number 2 not working
FIX - LogLevels.config issues with installer
FIX - Server crash with MORE line activity
FIX - call log search for names with apostrophe fails
FIX - Ringing handset problem with late provider SDP.
FIX - SkypeTx Codec - inaccessible ringing call
FIX - Skype TX codec calls cannot be answered in some instances
FIX - Defaulting to XML serialisation for browser REST HTTP requests
FIX - Incorrect number of channels parameter in Opus sdp
FIX - Skype TX avatars not being loaded from automation
FIX - Error in web manager when adding skype service
FIX - Cannot create skype service in web manager
FIX - Build to incorporate latest common components with SDP fixes
FIX - Occasional call stuck on Skype device
FIX - Web manager skype REST calls are not using internal call headers
FIX - Server is not sending keepalives to AS2 clients
FIX - AudioServer disposal problems on disconnection
FIX - Log entry changes
FIX - prevent  audio server 2 connections updating the config db on every connection
FIX - Don't send or respond to AS2 keep-alives unless initialise has completed
FIX - remove NV9000 blank routes from reverse route check
FIX - Scripts missing from Dolphin branch due to IS project clone from Chinchilla
FIX - Build to incorporate positive custom payload number fix for skype and vx handsets
FIX - SkypeTx call added to onair queue disappears when hungup
FIX - SIP items missing from LogLevels config file
FIX - licence check fails with version truncation issue
FIX - Remove webmanager link to v3 client
FIX - Web manager web service requests need to flag themselves as internal
FIX - licence count of main and mini services can get swapped when services are added/updated during runtime
FIX - Winners list person.Name if available

Monday, 27 August 2018

PhoneBOX 4 Client 4.8.1.34

Changes since: PhoneBOX 4 Client 4.8.1.28

NEW - Finish Systembase codec implementation
NEW - Modify "Source Routing - Lines" view
NEW - Codec text logos for Systembase / Prima

FIX - Call screener view - Handset answer next icon always showing
FIX - Dispo / alert icon showing over the top of line number on active call details header
FIX - Devices Details duplicated intermittently
FIX - Cleanfeed slideout menu button missing on codec until a call has been made
FIX - Conference button not working
FIX - Add logo for Nica X codec
FIX - RTE - null reference with call screener view on dialing
FIX - Occasional non-switch to outgoing calls in popped call details

PhoneBOX (General) 3.8.1.32

Changes since: PhoneBOX (General) 3.8.1.30

NEW - Complete rewrite of SIP audio device handset class
NEW - Prevent router reconcillation from ever removing inputs and outputs automatically

FIX - Early media ReInvites that send Update causing dial failure
FIX - Parking out of conference on VX devices not working
FIX - Direct dialing from handsets  not working
FIX - Occasional no audio after transfer immediate between handset devices

PhoneBOX (General) 3.7.1.43

Changes since: PhoneBOX (General) 3.7.1.42

New client (4.7.1.47)

PhoneBOX 4 Client 4.7.1.47

Changes since: PhoneBOX 4 Client 4.7.1.43

FIX - Devices Details duplicated intermittently
FIX - Cleanfeed slideout menu button missing on codec until a call has been made
FIX - Multiple draws scheduled at same time, prize selection issue
FIX - RTE - null reference with call screener view on dialing

Tuesday, 21 August 2018

PhoneBOX (General) 3.7.1.42

Changes since: PhoneBOX (General) 3.7.1.36

FIX - Wrong binding redirect for Microsoft.Owin

PhoneBOX 4 Client 4.7.1.43

Changes since: PhoneBOX 4 Client 4.7.1.36

FIX - Memory leak on switch converter case on call log gender
FIX - Null reference with call screener view on dialing
FIX - Occasional non-switch to outgoing calls in popped call details
FIX - Device slideout button click through causing call transfer to other TBU
FIX - Hangup slideout button not working on devices or destinations for telco calls
FIX - Problem with codec list not displaying after view change
FIX - Codec slideout button bar not working
FIX - Log entry error with view model cache manager

Friday, 27 July 2018

Virtual Director 2.8.1.26

Changes since : 2.8.1.9 (Beta)

New - Control BlackMagic preview bus independantly of program output via triggers.
New - Control On/Off air transition behaviour via triggers.
New - Control BlackMagic program and preview sources via triggers.
New - Studio Configuration screen redesigned.
New - Add Users in web manager.
New - Delete video in web manager.

[General]
Fix - Change of show from PB4 not working.

[Server]
Fix - Reported disk space available error.
Fix - Oasis publish queue export fails.
Fix - New video available  message not being sent to PB4 clients.
Fix - Reduce size of thumbnails served via API call.(Prevents PB4 memory issues and VLC not playing)
Fix - Switcher not being notified of studio configuration changes.
Fix - Video ending in multiples of 10 seconds does not create last thumbnail. (Prevents errors in PB4 client if video is 10 seconds long)
Fix - Daily usage summary email always reports zero recordings.
Fix - Items posted to publish queues not appearing on screen.
Fix - Live streaming not working.
Fix - OnAir publish queue items not cleared from OASIS once displayed on screen.
Fix - Ensure Path Settings only checking default value.

[Switcher]
Fix - State and Mic Levels not showing in web manager.
Fix - Blackmagic reconnection error.
Fix - Media not uploaded to BlackMagic switcher following a reconnect.

[Streamer]
Fix - Memory Leak following a recording.
Fix - Recording path location should not be case sensitive.
Fix - Streamer not creating recording folder if it doesnt exist.
Fix - Should not use server disk space values when checking whether to record.
Fix - File Transfer fixes.

[Web Manager]
Fix - Branding page does not refresh after cloning a brand.
Fix - Refactor video playback from web manager. (inc additional logging in web manager).
Fix - Automatic creation of inverse events in triggers. eg OnAir / OffAir
Fix - When editing a Mic Active GPIO the UI does not show the selected mic.
Fix - Correct spelling of "Yellobrik"
Fix - Keep Station / Show list in view during drag/drop operation.
Fix - Edit Show.
Fix - Squeeze parameters not shown when editing squeeze trigger.