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 - 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
