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PhoneBOX (General) 3.10.0.73 (BETA/FOX)

Changes since: PhoneBOX (General) 3.9.1.25

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