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
