Bionic Social 1.12.0.260

Changes since 1.12.0.258

FIX - Ingested messages are duplicating in message and message attachments tables

FIX - OnAir state updates not propogating

Bionic Studio Core / Portal 1.12.0.350

Changes since 1.12.0.347

FIX - Populate locations table with locations.bin file

Bionic Talkshow 1.12.0.345

Changes since 1.12.0.342

FIX - Add location lookup

FIX - Cannot create Anywhere invite without SMTP server configured

FIX - MoH failure for web client headset answered Anywhere cals

FIX - Temporary RTP port value changed from 16400 to 0 to solve audio loss on WPF client softphone calls

Bionic Studio 5.12.0.776

Changes since 5.12.0.772

FIX - Webclient showing show name instead of machine name as chat default

FIX - WPF/Webclient call state sync issue

FIX - Point formatting on the line

FIX - Last 30 minutes of chat not loading into werbclient on startup

FIX - Webclient show default problem

FIX - Anywhere Invite requests not adding reliably to Core DB

FIX - Changing view causing unexpected call routing

Bionic Studio Installer 3.2.0.61

Changes since 3.2.0.55

NEW - Copy locations.bin file for inclusion in CORE area lookup dataload

NEW - Installer is now signed

FIX - Correct bad version number

Camera One 2.12.0.62 (BETA)

Changes since 2.12.0.61

FIX - Occasional issues deleting recordings

FIX - LWCP (Axia QOR) audio sources do not reconnect if they are moved around on the desk

Camera One 2.12.0.61 (BETA)

Changes since 2.12.0.60

FIX - New recordings are not displayed if a previous corrupted recording causes the library load to fail on startup

Camera One 2.12.0.60 (BETA)

Changes since 2.12.0.59

FIX - Record trigger from RØDECaster Pro 2 not working when set to "When RØDECaster is recording" 

Camera One 2.12.0.59 (BETA)

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

PhoneBOX (General) 3.11.1.79 (STABLE/FROG)

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- Add None and Recorder types to extensions output selection

NEW - Optimise DNS lookups for SIP Hosts & Proxies

NEW - Allow override of Axia GPIO Node / LWRP port number

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

SkypeTX 1.11.1.36 (STABLE/FROG)

Changes since: 1.11.1.9

 

NEW - Add NDI holding source to maintain framerate when channel is idle

NEW - Add resolution 1080p25

Audio Server 2.11.1.68 (STABLE/FROG)

Changes since: 2.11.1.16

 NEW - Audio server update to retry if its a new version is available and the download fails

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

PhoneBOX 4 Client 4.11.1.89 (STABLE/FROG)

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

Camera One 2.12.0.56 (BETA)

Changes since 2.12.0.55

FIX - Problems connecting to the ATEM Mini Extreme ISO