Wednesday 24 January 2024

PhoneBOX (General) (STABLE/FROG)

Changes since:

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- Add None and Recorder types to extensions output selection

NEW - Optimise DNS lookups for SIP Hosts & Proxies

NEW - Allow override of Axia GPIO Node / LWRP port number

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 - 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