Changes since 2.12.0.56
NEW - AutoRecord enable / disable trigger now available
FIX - Streaming capability incorrectly picked up for ATEM Mini Extreme
FIX - Recording max length warning pops up at wrong time and doesn't clear when recording ends
Changes since 2.12.0.56
NEW - AutoRecord enable / disable trigger now available
FIX - Streaming capability incorrectly picked up for ATEM Mini Extreme
FIX - Recording max length warning pops up at wrong time and doesn't clear when recording ends
Changes since: 3.11.1.17
NOTE – The SIP version is available in both 32bit and 64bit using different installers. When switching between 32bit and 64bit ensure that you uninstall and reinstall, so the application files are in the correct Program Files folder. Broadcast Bionics are recommending customers migrate their server to 64bit to benefit from additional memory resource this offers.
NEW - Enhancement to SIP stack to allow for dynamic hostname resolution of endpoints, and to support TLS in the future
NEW- File based Music On Hold
NEW - File based audio device input
NEW - Extension GPO for call dropped indication
NEW - Implement pulse turn off for single pulse GPO events such as Call Dropped indication and implement for ADAM interface
NEW - Improve server start-up if license has been updated
NEW - Purge Director Media table after 90 days
NEW - Allow split first name last name values in directory entries
NEW - Use RFC2833 for sending DTMF for Audio Server and Softphones instead of generated tone
NEW - Add OAuth flow to web manager
NEW - Persist maximum client count between server restarts
NEW - Play ringtone to SIP devices when the trunk does not support early media [file location beneath exe folder Audio\defaultringtone.wav]
NEW - Allow a SIP Auth to be shared between services when using registered trunk
NEW - Display Last Access time for machines in the web manager
NEW - Add control of service state for busy/forwarding to the REST API
NEW - Implement Syslog logging with [Options] SyslogAddress ini file entry
NEW - Improvements to LUCI Codec control
NEW - Option to allow SIP Registrations to use Auth user in the From/To Headers
NEW - Server Ini file entry [options] checkBackupBeforeStarting=1 to cause primary server to wait for backup shutdown before full starting
NEW - Setting pulse on a VX Server state output to periodically check the correct show is selected
NEW - Improvements to LUCI Codec control, including jitter buffer per call.
NEW - AllDirectoriesInLookup option to force all directories to be search for caller lookup
NEW - Internal transfer of calls between services
NEW - Move telephony web sockets from More External Interface to core application and improve real-time call events and API
NEW - Add TLS 1.2 support to SMTP sending
NEW - Extend directory lookup on new call to include 'other numbers' - INI file entry - [options] OtherNumbersInLookup=1
NEW - Place call over backup trunk if ServiceUnavailable response received to initial call attempt
NEW - Add active show to client list in Web Manager
NEW - INI entry to set VoIP device colour to any valid HTML colour - [options] VoipDeviceHtmlColour=
NEW - Implement dynamic Ember+ router destination labels
NEW - Implement secure connection for Pathfinder Core on port 9602
NEW - Improve REST API DeviceLayout list to include linked devices and codecs
FIX - Improve display of device usage during long dialling and prevent stuck calls that can occur during subsequent dialling attempts
FIX - Regression since 3.11.1.7 - trunk failovers may not
work on newly created service, or service that has never been manually
closed/opened.
FIX - Anywhere Websocket reconnections not working properly
FIX - Pulse not working on Axia GPIO
FIX - Improve handling of situations where OK and CANCEL
cross on the wire resulting in stuck calls
FIX - Incorrect baud rate used for NicaX codec
FIX - Invalid SEQ number when sending PRACK to Handset
Devices
FIX - Multiple auto answer extensions attempt to answer the
same call
FIX - Prevent direction attribute being added multiple times
to parsed SDP
FIX - Retrieving lists of Anywhere services can delay
service configuration
FIX - Change service ringing notification to only present call to single device if it’s to be auto answered (audio server)
FIX - Sip authentication fails if provider offers qop auth-int method
FIX - Audio server wont attempt to reconnect if there is a
problem loading its configuration and it has no devices
FIX - Regression - ringtone from early media no longer
working
FIX - Sip display names are not correctly escaped for
quotation marks and slash characters
FIX - Error preventing calls from ringing if service
notification extension configuration not valid
FIX - Error when manually adding audio router IO output in
web manager
FIX - VX upset if a call is answered at the point there is
no provider SDP
FIX - Ensure only IPv4 addresses are used for SIP hostname
lookup
FIX - Improve efficiency of SIP registrations and fix
provider incompatibilities
FIX - Issue parsing SIP contact headers with multiple
entries
FIX - Anywhere calls can become stuck
FIX - Axia Multicast GPO pulse issues
FIX - A new service notification call to a device would not
be triggered if the previous call left the device using a transfer immediate
FIX - Alter version number log entry to indicate 64 or 32
bit build
FIX - Anywhere calls rejected when Force Auth is enabled
FIX - Nonce count SIP authentication value should be lower
case
FIX - Ensure directory call information sent from the client is used during call lookup process
FIX - Axia GPIO external interface unable to connect to Livewire virtual soundcard driver
FIX - Error when parsing SDP with unspecified video format
FIX - Improve Contact header parsing for expiry time when
registration messages contain multiple headers
FIX - Memory leak when during high periods of Director
messaging
FIX - Reintroduce SIP ptime value for default 20ms timings
and ensure it only appears once in each SDP
FIX - SIP check to prevent parallel re-invites not always working
FIX - VX Codec commands retry on timeout
FIX - VX failover triggered by connection events from other external interfaces
FIX - Call point not set when calling from a message
FIX - Improve log file rotation and purging
FIX - Inbound capacity changes not applied until a new call
arrives
FIX - Codec call length not correctly set for inbound calls
FIX - Email send causing delays at startup
FIX - REST API error when returning Line Layouts for show
FIX - Sip failover with authentication using incorrect user
name
FIX - VX show switching happening after startup before
extensions are ready to accept registrations
FIX - Ensure directory call details take priority when
calling from directory
FIX - Implement new Advantech ADAM6066 interface code to ensure connection if device becomes available after start-up
FIX - Improvements to External Interface keepalive / reconnection logic
FIX - Ensure backup server detects primary is active after
one successful ping
FIX - Ember+ indexing issues when using wildcard in path
FIX - Websocket server leak when publishing live call/chat data
FIX - Incorrect SDP session ID when responding to a re-invite request
FIX - Xnode issue
where wildcard version response was blocking future unsolicited messages
FIX - Advantech GPO not correctly triggering
FIX - Comrex codec unable to connect through Socks proxy
FIX - Corrupt ringtone audio played to RTP calls
FIX - Performance issues when under heavy inbound call volumes when using ringing
handsets
FIX - Ringtone audio played by the server not heard on VX
device
FIX - Web Manager - add pulse column to External Interface
Outputs list
FIX - Add domain to Comrex SIP calls
FIX - Alter Axia console label command for UK firmware
FIX - Comex codecs improvements for the software codec and
socks connection
FIX – Unable to configure Anywhere if only TLS 1.2 available
FIX - Unable to transfer calls out of SIP conference to another
device
FIX - Backup server not starting if licence is re-requested
FIX - SIP Registration failing due to case sensitivity
issues with header processing
FIX - Incorrect compare of SDP sending an unnecessary
REINVITE to handset devices
FIX - Advantech GPIO not pulsing correctly
FIX - Unable to set Next state of call that had been
internally transferred between services
FIX - Invalid data in Skype Account token can prevent account from logging in when a valid token is provided
FIX - Duplicate command to remove call sent to Audio Server
FIX - Ensure call is actually still on a device before
accepting a request to Park the call
FIX - Mayah codec info not displaying correctly in client
Changes since: 1.11.1.9
NEW - Add NDI holding source to maintain framerate when
channel is idle
NEW - Add resolution 1080p25
Changes since: 2.11.1.16
NEW - Add helper to detect default communications devices
NEW - Pass Anywhere WebRTC ICE status and call stats to
Talkshow
NEW - Support for ICE restarts for Anywhere reconnections
NEW - Music on hold audio provided by a SIP device
NEW - Enhance Anywhere call quality log with call ending
reason + session guid
NEW - Improve WAV file playback and implement for device
inputs with default path to audio folder
NEW - Allow devices with no physical sound output
NEW - Use RFC2833 for sending DTMF for Audio Server and
Softphones instead of generated tone
NEW- Update references to latest components and add Opus Ogg
file parser and conversion
NEW - Voicemail style device recording
FIX - Crash issue relating to Anywhere call stats
FIX - Ensure recordings that stop when they reach the
maximum length are made available to clients
FIX - Anywhere connection object not releasing resources
causing memory leak
FIX - If no path is specified for file MOH then use default
audio folder
FIX - Jitter buffer losing packets on Seq rollover
FIX - On marker bit received our packet interval should be
recalculated and data transfer / buffer size operations adjusted accordingly
should it change
FIX - RTP Sequence number not correctly rolling over and
always starting from 0
FIX - Improvements to WDM device reliability to prevent of
crashing at start of call and allow audio to continue if device is disconnected
and reconnected.
FIX - Call recording not working in Automatic mode
FIX - No caller audio after very short ringing period
FIX - Improve log file rotation and purging
FIX - Prevent crash caused by RTP decoder
FIX - Multicast group rejoin/join issues with rapid fire
park cycles
FIX - Corrupt ringtone audio played to RTP calls
FIX - Logged error if File Transfer port hit by scanner
FIX - Problem with caller audio if packet marker with no
payload received
FIX - Remove timeout causing delays when calls are removed
from device/hold
FIX - Unable to reconnect to server after keepalive timeout
Changes since: 4.11.1.13
NEW - Improvements to MCR devices view - fixes, UI changes, add transfer and dial options to devices, prevent device selection, change device layout, add DTMF slide out, display any screened device as pink.
NEW - Improve usability of WhatsApp Audio
NEW- Implement media attachments to Facebook Messenger messages
NEW - Allow split first name last name in directory entries (requires server update), add secret mode dialling, and make call button use consistent with call log.
NEW - Use RFC2833 for sending DTMF for Audio Server and Softphones instead of generated tone
NEW - Allow clients using separate shows to shared the same next device indication (requires show guid as chat group name)
NEW- Alter codec only view to also display telco devices
NEW - Close dial pad if device page is changed
NEW - Add all non secret dialled numbers to the last 5 list on the dialpad
NEW - Allow directory entries to be added to the OnAir queue
NEW - Prevent duplicate global directory entries for the same number
NEW - Support for dynamic social accounts and Message Log filtering by Social Account Group
NEW - Allow source list grouping sort order by prefixing the name with num_
NEW - Add client based Axia GPIO that can trigger when Softphone has an active call
NEW - Encode WhatsApp audio to WAV
NEW - Hide person related call details fields in MCR view
NEW - Ini file [options] UseSoftwareRenderMode=1 to bypass hardware graphics rendering
NEW - Add Director error message when failing to connect to remote studio
NEW - Add server availability indicator to MCR devices view
NEW - Add dial-pad style buttons to Transfer and Forward popups
NEW - Improvements to Directory including duplicate number in check in the 'other numbers', dialling from other numbers and hide codec section if no codecs configured
NEW - Move WhatsApp voice files to recording folder
NEW - Show Director current show in change show dialog.
FIX - Prevent switching line pages/tabs sending unnecessary refresh requests to the server
FIX - Regression in 4.11.1.13 - license exceeded message no longer displayed in client splash screen
FIX - Calls in the 'Ready' queue in Classic Presenter view take up all available space
FIX - Handset sometimes shows as unavailable after view change, even though calls can still be made
FIX - Point text can be limited to one line event though there is more available space in certain line sizes
FIX - Tags and Screened tick box not fully visible in Call Screener (3 Column) view.
FIX - Screened state could become unset after several unpark operations
FIX - Selected message in the presenter viewer should be updated immediately if the content changes
FIX - Unable to respond to WhatsApp messages
FIX - On air queue content not correct after view changes
FIX - Popped call log item text not wrapped if call details are displayed in screener section of view
FIX - Regression - ringtone from early media no longer working
FIX - Prioritise SMS accounts for message sending
FIX - Adjust number of lines per row if >2 columns are used to ensure they are evenly balanced
FIX - Change layout of prize display so all prizes are visible without scrolling
FIX - Docked bar space not released when client disconnected from server
FIX - Tag values not always displayed correctly in the On Air Queue
FIX - Update Active Directory components to latest version
FIX - WhatsApp replies appearing as SMS messages
FIX - Include defaultringtone.wav with the client installer
FIX - No ringtone on VoIP softphone when trunk early media is not present
FIX - Audio library changes to prevent softphone crashes
FIX - Dragging between devices can move active device selection
FIX - Improve screen reader interoperability with the selection sequence at startup
FIX - Secret mode dialling not reliably working when calling from the Directory
FIX - Correct French spelling in client title bar
FIX - Client disconnecting during start-up routine if configured with a large number of devices and lines
FIX - Client slow to start on some enterprise networks
FIX - Hide caller name on lines for secret calls
FIX - Improve usability of WhatsApp audioclips
FIX - Prevent error if social server returns empty image data
FIX - Three column classic view does not show device pages
FIX - Add option to resize slideout control when buttons change
FIX - Allow colouring of device tabs
FIX - Directory filter button not shown as latched when active
FIX - Lastname not included in call log filter
FIX - No caller audio after very short ringing period
FIX - Unable to dial a skype call from the call log on a handset when not in the skype contact list
FIX - Allow VoIP headset to be the default skype device
FIX - Resize line slideout control when buttons change
FIX - Softphone issues decoding first packet of segment and determining updated packet duration
FIX - Allow codecs to be hung-up whilst ringing in
FIX - Source name group displayed on codec selection buttons
FIX - Trap some errors relating to transfer and phonecallrecordchange.
FIX - Call details tabs changing on data entry
FIX - Enable click to dial from OnAir queue as default for MCR view
FIX - Enhance call details tab to hide Info tab on new call records and provide basic detail tab for MCR view
FIX - Improve display of bitrates on codecs
FIX - Transfer popup being cut off when device close to the bottom of screen
FIX - Answer Key device selection incorrectly applied to all answer operations
FIX - Call details not populating from directory when using the second number
FIX - Include time of WhatsApp voicenote in the filename
FIX - Fader icon missing from destinations
FIX - Prevent crashes when double clicking the slide out menus
FIX - Crash if codec dial pad left open for a long time
FIX - Improvements to codec dial pad, mostly for Comrex codecs
FIX - Reduce debounce restriction when adding calls into conference
FIX - Unable to drag calls from conference using a line in SIP systems
FIX - Ringtone issues when audio begins quickly
FIX - Unable to create directory entry from call log item if number 2 field empty
FIX - Crash when viewing some Facebook videos
FIX - When switching to a remote camera UI briefly shows incorrectly selected local camera
FIX - VoIP handset colour not following server config
Changes since 2.12.0.55
FIX - Problems connecting to the ATEM Mini Extreme ISO
Changes since 2.12.0.46
NEW - Triggers can now be configured to switch cameras
NEW - A switcher input can now be assigned as the mute source
NEW - Mute graphic can now be set as disabled
NEW - Triggers tab now includes live indicators
NEW - It is now possible to limit the number of CPU cores utilised for clip creation
NEW - A recording duration limit can now be set
NEW - Improved workflow when deleting an audio interface
NEW - Audio input selection enhanced with new audio level picker
NEW - Date picker now offers more than the previous week if recording purge is set appropriately
NEW - Audio config tab improved by displaying audio interfaces by type
NEW - Live preview now has rounded corners
NEW - Improve indication when recordings are being processed
NEW - Streaming fields now have a clear button
NEW - ATEM streaming option is now hidden when using the base model ATEM Mini
NEW - ISO recording option is now hidden if the ATEM Mini is connected over USB
NEW - There is now a manual switch option for switcher inputs (making it visible despite no mic being assigned)
NEW - Improve labelling of switcher triggers
FIX - Sticky header and footer fixes
FIX - Improved handling of disc being full
FIX - ADAM 6066 interface reconnection improvements
FIX - Timeout now handled properly when adding ADAM 6066
FIX - Clicking the mic icon next to an audio interface now selects that interface in the pop up
FIX - Clicking the mic icon on an audio device doesn't now show a threshold
FIX - Inputs tab elements now have consistent horizonal alignment
FIX - Missing Ember+ DLLs are now included
FIX - Config file not being updated to reflect mute and bug graphics
FIX - It's no longer possible to add an LWRP source with an invalid IP address
FIX - Layout problem if no mute graphic is set
FIX - Trigger level for Wheatstone input incorrectly saved
FIX - Updating switcher input name doesn't always refresh UI
FIX - Obsolete -frames ffmpeg option no longer used
FIX - Recording unable to start with ATEM Mini Pro (non ISO) model
FIX - Microphone triggers now ordered the sane as camera inputs
Changes since 2.12.0.38
NEW - It is now possible to add multiple instances of the same interface type
FIX - Invalid pixel formats are now converted to 32bit when uploading mute and bug graphics
FIX - RodeCasterMIDI faders were erroneously receiving AxiaGPIO events
FIX - Axia and Wheatstone GPI pins were being incorrectly displayed
FIX - SLIO event numbers were being incorrectly displayed
FIX - Video and audio capture device names overrides were being ignored
FIX - Audio interfaces with the same channel names were causing issues
FIX - Audio and GPIO interfaces of the same types must have unique IPAddresses
FIX - RodeCaster ASIO driver check improved by using registry keys
FIX - Wheatstone audio failing to start on startup
FIX - Wheatstone setting incorrect trigger values
FIX - Wheatstone SLIO query fails to respond when Wheatstone audio level connection is being set up
Changes since 2.0.0.71
NEW - EU Server for European Customers (GDPR).
FIX - Allow + in email addresses (google feature).
FIX - Dial pad cursor issues.
FIX - Show number on lines if no name after matching location.
Changes since 2.12.0.32
NEW - Support for RØDECaster Duo
FIX - FTP download does not recover if disk is removed.
FIX - RØDECaster camera switches are not disabled when AutoSwitch is active
FIX - Triggering stream start from a RØDECaster pad is causing hanging if no stream is set up
FIX - Recording not working on ATEM Mini Pro when connected via IP
FIX - RØDECaster ASIO driver not identified once it's been installed from link
FIX - Rogue PNG files preventing FTP folder delete
FIX - Connection error when connecting to ATEM Mini over IP when config already exists
FIX - Recordings stopping when a RØDECaster fader is closed despite others still being open