Tuesday 25 June 2019

PhoneBOX (General) (BETA/FOX)

Changes since: PhoneBOX (General)

NEW - Implement new mechanisms for dealing with call log records when server and client are in different timezones
NEW - Add custom fields support
NEW - Add configurable anywhere emails per station
NEW - Add show based setting to determine if child message cause the base social item to jump to the top of the message queue
NEW - Improvements to Luci codec implementation
NEW - Implement Prodys Quantum codec type
NEW - Allow devices to be paged like lines
NEW - Add capacity for restricting access to client views in device layout
NEW - Configurable Jitter buffer setting for each service
NEW - Dual name field - Real Name & Display name option
NEW - Support fader strip labels with caller name on Axia consoles without UK firmware
NEW - Change chat database times to UTC to ensure time display is correct on clients in different timezones
NEW - Allow web sockets publish queue to work with Skype devices
NEW - Add "video enabled" flag to web sockets publish queue
NEW - ExtensionInfoUpdate in More external interface should attach fader up status
NEW - Add caller prize info to API to allow query of 3rd party CRM
NEW - Add call API should provide the option to specify a point
NEW - Add mechanism to send Skype avatar to Skype codec clients
NEW - Add a special chat code that will send a message to all connected clients
NEW - Disable audio processing on Skype TX calls
NEW - Send relayed now playing information to anywhere for relevant shows
NEW - Application command line parameters to override configured values
NEW - Send on air queue data to anywhere server
NEW - Aeta extract sip number and name
NEW - Implement client edge on REST api
NEW - Build with latest BBCommon

FIX - Comrex codec password failure causes loop of retries draining resources
FIX - UltiDev preqrequisite being downloaded from web unnecessarily
FIX - Anywhere webportal chat messages are not shown (sent or receiving)
FIX - Dial requests from Fusion Console switcher were ignored
FIX - Build to include new client
FIX - E164 / Enhanced number format support for US locale numbers
FIX - VSset caller id appears as sip URI when no name set
FIX - Call log Wildcard search does not return expected result when using a foreign language
FIX - Location lookup broken - Anywhere refactor
FIX - Lookup method error with webhookResult causing VX and IPO call control to fail
FIX - Call log entries not appearing reliably
FIX - Skype calls showing Skype name instead of full name on lines, devices and codecs
FIX - SkypeTx line with video enabled from a hybrid is not updated when transferred to a handset
FIX - Fixes to some async calls to Skype TX automation component
FIX - Anywhere on air queue messages reading social media type from incorrect field
FIX - Skype TX Devices do not inherit through appended device layouts
FIX - Do not fail a sip device call with early media if UPDATE response is 491 - Request Pending
FIX - Upon attempting to create a custom field via PM2 webmanager a SQL error occurs
FIX - Improvement to LUCI codec SIP operation for version 5.0.29
FIX - Webhook drop call only triggered by remote end, and add new fields to responses
FIX - Skype codec shows video option on slideout even when no video configured
FIX - Arabic names are not searchable in Call Log Search
FIX - Improve performance of previous call lookup
FIX - Disable Skype audio processing for codec devices
FIX - Problem with Proxy Authorization on PRACK messages
FIX - Send chat group name to anywhere on connect
FIX - Ensure a call log entry is created if webhook lookup fails
FIX - Page name field nulls in device layout codec table causing codecs not to load in client
FIX - Aeta -  call log issues for incoming sip calls
FIX - Handset conference stuck when last call removed
FIX - Audio device handset exception thrown on hangup
FIX - Stuck call on handset device if call cancelled while ringing out
FIX - Fix anywhere native SIP handset call
FIX - PB Vx won't show calls if the Vx config is not supported
FIX - Sdp with Ack causing answering / handset re-invite deadlock and subsequent answer delay
FIX - Call stuck on handset devices due to provider drop before handset session establish
FIX - Voip handset not shown if other optional devices are set in the layout but not selected by the user
FIX - Rebuild with Sip stack including REFER auth header fix
FIX - Exceptions during answering incoming provider call to sip devices causes stuck calls