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