PhoneBOX (General) 3.11.1.99 (BETA/FROG)

Changes since: 3.11.1.79

NEW - Implement Probel SWP-08 routing protocol

NEW - Allow sip domain to be specified in Failover hosts

NEW - Allow configuration of presented CLI that is different from SIP username.  (use | in SIP username field)

NEW - Experimental - SIP TLS - enabled when local SIP port is 0

NEW - Replicate service state to backup server

NEW - Add ability to select call format when dialing from the API

NEW - Process Diversion header to learn source of a forwarded call

NEW - Include Working Set memory usage in minutely log entry

NEW - Tieline dialpad changes to support Tielink destinations and audio profiles

NEW - Improve Comrex and Luci call quality indication

FIX - Internal code change to SkypeTX Codec to improve event hooking

FIX - UPDATE message during established call can cause failure

FIX - Unable to parse IPv6 addresses in SIP headers

FIX - Ensure SIP digest authentication is correctly refreshed once stale

FIX - Direction SDP attribute not compared causing inactive calls to fail

FIX - Prevent SDP with Crypto being passed to handsets to avoid calls starting with SRTP

FIX - Trap recursive device layouts from causing crash in API DeviceLayout/List

FIX - Update third party libraries

FIX - Unable to answer call when SDP contains duplicate entries

FIX - Repeating CNONCE value in SIP Authentication causing anti replay detection to terminate session

FIX - UPDATE message during inbound ringing call can cause failure

FIX - Improvements to real time websocket stream

FIX - Prevent country code manipulation of inbound calls when EnhancedNumber processing is enabled

FIX - Service creation broken by service state replication change

FIX - Comrex delay indication incorrect

FIX - Comrex profile to use CBR and new jitter settings

FIX - Call details not available for all types of Comrex codec calls

FIX - Enhance Tieline codec to report SIP call information

PhoneBOX (General) 3.11.1.79 (STABLE/FROG)

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- 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 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

PhoneBOX (General) 3.11.1.17 (STABLE/FROG)

Changes since: 3.11.1.1

NEW - Add new extension output actions to trigger GPO when primary or backup servers are active

NEW - Music on hold audio provided by a SIP device

NEW - Allow multiple channels of Skype video to be used simultaneously

NEW - Allow configurable SIP proxy override

NEW - Configurable connection timeout for websocket client (5 second default)

NEW - Add new external interface type for Telos VX and provide show switching caperbilities

NEW - Prevent display of passwords in the configuration web UI

NEW - Change licencing endpoints to use HTTPS

NEW - Add new views for vertical devices, and alternate source routing views

NEW - Indication when lines are Out of Service, includes Anywhere and SIP registered extensions

NEW - Email alerts when trunks go in and out of service - server.ini [email] trunkStateTo=address

NEW - Default anywhere service to using web sockets in config form, and ensure service is correctly set when saving initial entry

NEW - Improve websocket logging

NEW - Improve logging for Virtual Director interface

NEW - Add trunkOos (trunk out of service) property to Service REST API response

NEW - Internal changes to support Anywhere call state and stats display

NEW - Reduce number of worker threads to save memory usage, can be adjusted with ini file [options] minWorkerThreads entry

FIX - Incoming MoH device registrations not working

FIX - Pathfinder Core virtual router IO can stop working after package IO is modified

FIX - Websocket connections stuck trying to send in an aborted state

FIX - Session ids and versions are not generated relative to epoch time

FIX - Unable to negotiate SDP where the same media format is defined more than once

FIX - Virtual director interface unable to parse version in some regions

FIX - CANCEL message sent to end an outbound call before answer is not compliant with SIP specification

FIX - Regression with Ack being sent without Record-Route headers

FIX - Add HTTP response code to websocket client error log entry

FIX - Anywhere account setup failing to use configured web proxy in Web Manager

FIX - Optimise memory usage of Person record cache

FIX - Tieline codec reconnection attempt on shutdown

FIX - Error when starting server with an active call on a codec

FIX - Error when checking availability of SIP Servers

FIX - Do not include Anywhere service licences in total counts

FIX - Aeta codec not sending decoder changes messages to client

FIX - Sip domain with port incorrectly overriding the destination sip port

FIX - Call stuck on handset device after second failed answer attempt

FIX - Trap error if call terminates before lookup completes

FIX - Verbose SQL logging duplicating some log entries, and not substituting all parameter values

FIX - Change end of call database functions to be asynchronous and ensure the call length is updated after the initial lookup completes

FIX - Sip performance improvements relating to thread waits and sequencing of messages

FIX - Handset dialling problem with trunk early media causing no audio

PhoneBOX (General) 3.11.1.1 (STABLE/FROG)

Changes since: 3.11.0.29

NEW - Allow configured divert number to be used for dial gpi

NEW - Secure client connection data transfer with TLS 

NEW - Change SDP library to separate processing of audio and video media segments

NEW - Allow ember gpio to use node wildcard at end of path

NEW - Update Locations.bin with Lithuanian Area Codes

NEW - Rebuild with PM2 build containing 'Winners' report filtering fix

NEW - Update Bionics Director Camera list when camera image added in web manager.

FIX - Handset devices using incorrect Skype Media Server IP address

FIX - Change safety code to hangup handset if provider call ends during setup, so that it uses the full Stop Audio process

FIX - Optimise memory usage of Person record cache

FIX - Unable to answer calls after terminating calls using the handset

FIX - Change to SIP header processing to prevent errors resulting in stuck handset calls

FIX - Prevent shortcodes from automatically being stripped from inbound calls - strip number prefix can be used for this purpose

FIX - Alter source label parameter name used in Axia Consoles router labels

FIX - Prevent ringing continuing after initial response has been sent

FIX - Ensure a SIP BYE message is sent after a CANCEL if it has crossed over with the remote end accepting the call, including before RINGING received

FIX - Call stuck on handset devices when remote hangup occurs during establish

FIX - Websocket client connection object not releasing pipe threads

FIX - Modify invite/reinvite moments to ensure video segments of SDP are maintained

FIX - In VX do not show all services as busy when Busy All is active

FIX - Problem with multiple notification handsets not all being able to answer calls

FIX - Anywhere lookup of contact from PB server fails if "Name Format" is set to "First name & Surname"

FIX - Fixes to purge routine to prevent person records linked to Anywhere Invitation from being removed before the Invitation expired, removal of expired Invitations in the database, and solve timeouts relating to call and person index maintenance timeouts

FIX - Improvements to stability of hansdet/vx devices where calls are cleared during answer

FIX - Changes to SIP CANCEL handler for conditions where calls are cancelled during setup

FIX - Optimise webhook config lookups to use a cache rather than query database for each call

FIX - Additional logging for errors relating to on air queue management of inactive phone calls

FIX - Softphones failing to register with wrong realm value

FIX - Issues with automatic rejection of banned callers

PhoneBOX (General) 3.11.0.29 (BETA/FROG)

Changes since: 3.10.1.66

NEW - Improve Anywhere invitations to allow customisation of email from field and send over SMS

NEW - Allow a default source to be routed to codecs when they are un-routed (uses the reverse route selection)

NEW - Sip stack performance improvements in request processing

NEW - Change Anywhere call signalling flow for moves between audio servers

NEW - New view: Self Op - No Messages

NEW - Implement new Ember GPIO functionality with configurable paths

NEW - Implement new code for Axia Gpio interface

NEW - Implement Tieline codec dialling and improved status

NEW - Implement phone number parsing in common codecs

NEW - Improvements to client command processing

NEW - Default mode to use new Common codecs

NEW - Build with SkypeTxAutomation 2.21.309.1 for JToken memory growth fix

FIX - Reconnect on idle setting missing from Machine config entries on SIP systems

FIX - Problems using handset after dialling directly

FIX - Caller name not updating for Anywhere OAQ for active calls 

FIX - OAQ replace operations aren't repeating to Anywhere 

FIX - Audio server check licence failure after new audio server added

FIX - Enhance Purge  routine to manage VirtualDirectorLinks table and reorganise person and phonecall indexes

FIX - Prevent crash when changing an Axia Console External interface that isnt connected

FIX - Prevent stuck call when call setup received a Request Terminated response

FIX - Anywhere OAQ does not update caller details when they are changed on the PB4 client

FIX - Skype account no longer retrying logins after initial failure

FIX - Anywhere calls dropped while on hold when answered on a different AS get stuck

FIX - Stop writing to VirtualDirectorLinks table that is no longer used

FIX - Exception during Anywhere person lookup request

FIX - Check for | character in Route names sent to fusion and strip right hand side

FIX - DHD reconnection caused server crash

FIX - Improve robustness of conversion of numbers to dialable format

FIX - Fixes to support SDP parser changes in SIP library

PhoneBOX (General) 3.10.1.66 (STABLE/FOX)

Changes since: 3.10.1.49

NEW - Allow a default source to be routed to codecs when they are un-routed (uses the reverse route selection)

NEW - Implement Tieline codec dialling and improved status

NEW - Certain Anywhere calls not using proxy

NEW - Automatically include video in Skype Calls on devices with names prefixed with *

NEW - Default mode to use new Common codecs

NEW - Build with SkypeTxAutomation 2.21.309.1 for JToken memory growth fix

NEW - PM2 build

NEW - Remove Anywhere branded header from Anywhere emails

FIX - Problems using handset after dialling directly

FIX - Audio server check licence failure after new audio server added

FIX - Enhance Purge  routine to manage VirtualDirectorLinks table and reorganise person and phonecall indexes

FIX - Prevent crash when changing an Axia Console External interface that isnt connected

FIX - Prevent stuck call when call setup received a Request Terminated response

FIX - Anywhere OAQ does not update caller details when they are changed on the PB4 client

FIX - Skype account no longer retrying logins after initial failure

FIX - Exception during Anywhere person lookup request

FIX - Prevent Anywhere services from consuming service licences

FIX - Database error can prevent Skype accounts from initialising

FIX - Skype automation memory management changes

FIX - Stuck handset device call if two clients unpark to the same device

FIX - Remove attached image from Anywhere invite email

FIX - Anywhere answer fails when no m line present in sdp

FIX - Sdp parsing error in commin library (fox only)

FIX - Element / Fusion interface not working with Quasar console

FIX - Audio SDP parsing needs to use the connection line linked to the audio media line

FIX - Banned voicemail check for incoming SIP call needs to refuse the call with BusyEverywhere

FIX - Clients unable to start if show directory converted to be global

PhoneBOX (General) 3.10.1.49 (STABLE/FOX)

Changes since: 3.10.1.33

NEW - Allow Mayah codec to function in SIP mode

NEW - Comrex codec read sip username and set as logged in user for idle text display

NEW - Server ini file option to override default worker thread counts

FIX - Transferring calls from audio server devices can result in failed transfer

FIX - Audioserver devices reporting offline when multiple connections overlap

FIX - Case sensitivity issue when using Audio Server MachineName override

FIX - SIP CANCEL can run during call setup leaving problems behind

FIX - Change Mayah codec message processing to be more thread efficient

FIX - Log local UDP port used for level information sent to clients

FIX - Backup server offline can cause delay to OAQ updates at end of call

FIX - Improvements to Skype reliability when accounts have problems logging in

FIX - Unable to make outbound calls with audio server devices

FIX - Provider OPTIONS pings not working when services use different local port to the main system port

FIX - Transfer operation not locked from overlapping requests as per Unpark and Answer

FIX - Make SIP device/handset hangup routine return success/fail and route call operation code accordingly

FIX - Repeated websocket errors in server log

FIX - Skype media server connection status causing startup service logins not to happen

FIX - SIP Prack message not compliant with some providers

FIX - Improve Skype handling of failed call setup scenarios in Lines mode

FIX - Improvements to Handset to prevent failed calls getting stuck and then causing answering of inbound calls to fail

FIX - Optimise Mayah codec thread usage

FIX - Notify extensions can stop ringing if calls drop during setup of ringing

FIX - Directory person lookup not including the second phone number field

FIX - Keepalives and telco messages stop being processed by the server

FIX - Crash inside SkypeImage Finalizer

FIX - Optimise Skype Avatar image handling to improve RAM usage when there are large number of contacts and accounts

FIX - A single Skype Channel failure marks all devices on that server as off line

FIX - Server loading MOH config several times on startup

FIX - Problems calling Skype For Business search result that is not a contact

FIX - Skype Codec stuck call after failed dial

PhoneBOX (General) 3.10.1.33 (STABLE/FOX)

Changes since: 3.10.1.18

NEW - Reduce Anywhere signalling for call moves within an audio server
NEW - Limit client TCP data retries to 3 and allow option to override number of retries
NEW - Change client level meter updates to be sent using UDP
NEW - Add support for anywhere person information lookup requests
NEW - Update Bionics Director Camera list when camera image added in web manager.

FIX - stuck handset calls related to provider call drop during setup of handset call.
FIX - Livewire device recorders should not consume Audio Server device licences
FIX - Skype Device List in Device Layout not sorted correctly
FIX - Instabilities caused by client connection problems when receiving level meter data - no free pool threads
FIX - Updates to inactive OAQ calls not replicated to backup server
FIX - Enhanced number lookup applied to numbers shorter than the minimum lookup length
FIX - Prevent mismatch of Skype Channel Count from allowing media server from initialising
FIX - '#' sign in dial prefixes causing SIP calls to fail
FIX - Add audit entry for Skype Contact Add and Removal
FIX - OAQ Skype calls not converted correctly to their inactive state at end of call
FIX - Unable to park/unpark Skype call after server restart
FIX - Unable to add Skype For Business Contacts
FIX - Issues with SkypeTX Codec including black codec on disconnect and person record data inheritance
FIX - stuck handset calls related to provider call drop during setup of handset call.
FIX - Auto allocated devices are not always un-routed if the call fails or drops
FIX - Prevent problems caused by Denial of Service attack on Comrex Codecs
FIX - Calls stuck in an uncontrollable state if their host device's registration times out
FIX - Registration expiry mechanism for handset devices can be affected by DST changes
FIX - Prevent flapping Skype Device state if one channel of a media server is not available
FIX - Change log level for Handset device register timeout to Warning
FIX - Updates to inactive OAQ calls not replicated to backup server
FIX - stuck handset calls related to provider call drop during setup of handset call.
FIX - Trap error that can cause Screened Held feature to stop working for a particular show
FIX - Prevent provider re-invites from provider causing loops with handset devices
FIX - Improve reliability of Skype call functions after failed call attempts
FIX - Problems with replication of on air queue items to backup servers
FIX - Improve handling of polling clients that dont full connect
FIX - Pool thread exhaustion with Anywhere services enabled
FIX - First / sporadic anywhere calls ring on two lines
FIX - Anywhere invites from the client that don't contain an email should display a different screen
FIX - Anywhere call on a locked line should not go to callback if terminated remotely
FIX - Client connections not processing concurrently as intended
FIX - Anywhere calls should not be able to join a conference
FIX - Skype ringtone not working on outbound calls
FIX - Pool thread exhaustion with Anywhere services enabled
FIX - Calls not correctly parked if hold source AudioServer is not available
FIX - Raise error message to client if call to handset device cannot be initialised when dialling
FIX - Prevent re-invites with the same key SDP components from starting and stopping audio server calls
FIX - Skype calls sometimes do not persist or can become uncontrollable through server restarts/fail overs
FIX - Ptime values were not honoured in SDP answer or passed to Audio Server
FIX - Prevent unnecessary re-invites to handset devices
FIX - Prevent re-invites with the same key SDP components from starting and stopping audio server calls
FIX - Send router commands on a queue so they dont delay telephony operations
FIX - Handset check to ensure that calls are correctly hung-up on the switch of device if the command sequence overlaps
FIX - Not all external interface types were correctly answering inbound calls through a remote trigger
FIX - Parking an Anywhere call two times will cause it to be stuck on hold.

PhoneBOX (General) 3.10.1.18 (STABLE/FOX)

Changes since: 3.10.1.4

NEW - Disable anywhere chat on backup server when not active
NEW - Timer to log back into default Skype account after a period of inactivity when a different account is logged in
NEW - Allow Audit Log Entries to be written to a separate file.

FIX - Fix websocket bug that prevented client connections from working correctly
FIX - Skype channels and devices can become locked out after certain dialling failures
FIX - Problems with calls being stuck on handset devices when providers clear down calls during setup
FIX - EnhancedNumber conversion not applied to outbound calls to ensure E164 numbers are stored in local format
FIX - Private/Secret call details not always restricted
FIX - Draw source error in some regional settings
FIX - Prevent unnecessary provider UPDATE during call setup on handset device
FIX - Comrex codec errors on call disconnect due to parallel processing of two status messages
FIX - Prevent SIP calls from being able to be dialled with an empty number
FIX - Errors when updating person records if OAQ contains no call records
FIX - Remote drop of SkypeTX codec call would not populate client call log
FIX - Ensure anywhere connections only happen on the online server in a backup/primary setup
FIX - Call lookups failing if records exist with a NULL StartTimeUtc
FIX - Prevent router destination fader change events being sent to all clients
FIX - Errors relating to SkypeTX service and channel actions
FIX - Installer upgrade to add Chat StartTimeUTC column creates extra replication batches
FIX - SDP with ACK not triggering after a REINVITE received without SDP from provider
FIX - PopulateStartTimesUTC Update process failing on installations using Sql Windows Auth
FIX - FATAL caused by AxiaGPIO interface re-connections
FIX - Ensure RestApiRequest is disposed in all places
FIX - Comrex codecs not automatically reconnecting if socks proxy host cannot be resolved
FIX - Display user friendly Skype name on SkypeTX Codecs
FIX - Service withheld anon prefix not applied to Callback calls
FIX - SkypeDevice stuck in onair state after failover scenario
FIX - Allow dynamic withheld to use anonymous or shortcode CLI restriction
FIX - E164 / Enhanced number format support for US locale numbers
FIX - Errors logged when blank line received from Axia Console External Interface
FIX - Build with common fix to web request disposal
FIX - HangUp GPI  does not enforce fader open or call state lock restriction
FIX - SkypeService DefaultAccount property not passed to client at startup causing incorrect login menu item to be displayed instead of logout
FIX - Skype services in studio based line layouts ignored

PhoneBOX (General) 3.10.1.4 (STABLE/FOX)

Changes since: 3.10.1.3

FIX - Issues with multiple users unparking the same call
FIX - Anywhere web service call delaying web manager  service configuration
FIX - Respone exceptions to softphone or sip handset reinvites don't unwire phonecall events
FIX - When ACK not received to OK when answering incoming calls to handsets, both legs should be dropped

PhoneBOX (General) 3.10.1.3 (STABLE/FOX)

Changes since: 3.10.1.2

FIX - Opus asymmetric payloads not updating in some SIP messages
FIX - ReInvites should remove unsupported codecs

PhoneBOX (General) 3.10.1.2 (STABLE/FOX)

Changes since: 3.10.0.82

NEW - Implement Axia Multicast GPIO external interface
NEW - Implement new VX Codec as part of new CommonCodec architecture - enable with server.ini [options] useCommonCodecs=1
NEW - Line Layout groups in configuration
NEW - Implement Audit logging of user actions
NEW - Allow Audio Server devices to auto answer if included as a service notification extension
NEW - Allow user to change login of Skype Services
NEW - Line Layout groups in configuration
NEW - Ensure HideSensitiveData system setting field to be sent to clients
NEW - Add Diversion header to SIP 302 Forwarding response
NEW - Implement version 2.19.1106.2 of SkypeTxAutomatoin (removes Dispatcher)
NEW - Allow SIP services to permanently restrict outbound CLI by placing the word 'anon' in the service shortcode dial prefix
NEW - Build with audio server jitter changes
NEW - Add new ExternalInterfaces common library to server installers
NEW - Populate FirstName and Surname fields in database for anywhere calls
NEW - Support SIP invite to withhold number
NEW - Integrate automation version 2.19.505.1
NEW - Multilevel Skype Device inheritance from linked layouts
NEW - Allow SIP services to permanently restrict outbound CLI by placing the word 'anon' in the service shortcode dial prefix
NEW - Call recordings to use host-name for retrieval by clients

FIX - Codec call log creation fixes
FIX - Pagename in DeviceLayoutCodecs does not accept NULL or blank value
FIX - UPDATES incorrectly sent when no Allows header present
FIX - DefaultDevice setting for Skype devices not being communicated to clients
FIX - SIP Services failing to de-register on shutdown
FIX - Some config tables not synced to secondary or included in the XML config export
FIX - Fixes and enhancements to Skype Services and dynamic login
FIX - Skype services in studio based line layouts ignored
FIX - Opus payload number changing unexpectedly during unpark operation
FIX - Anywhere chat name retrieval not working for full names
FIX - Change anywhere email timeout to 20s
FIX - Moh with various Opus payload numbers doesn't work
FIX - Prevent Skype Service logging in disabled accounts, trap errors at login if token is invalid, from preventing further services initialising, and improve logging
FIX - Shutdown not completing causing upgrades to get stuck and EXE needing to be manually killed in task manager
FIX - Fatal when logging out from Skype Service
FIX - AD config tables not purged prior to secondary configuration sync
FIX - Change the way Audio Servers are managed within the server to use host-names rather than IP addresses.  This prevents problems with IPs that cannot be resolved on server startup or may change during the time the server is running
FIX - Fatal in server with websocket / deserialization
FIX - Problems using SQL Windows Authentication following work to encrypt SQL Passwords
FIX - Build to fix locking issue when rest API unavailable to skype common library
FIX - Chat from Anywhere user is not correctly labelled with the sender's name
FIX - Clear local skype device object when clearing audio/video devices to prevent bad logging
FIX - Outbound calls that involve codec renegotiation on answer failing
FIX - Line Layout Group: Accessing assigned line layouts via page does not redirect correctly
FIX - Protect against Null To String conversion error in DeviceLayoutSkypeDeviceDto which caused api/v1/devicelayout/list call to fail.
FIX - Register SkypeDevices with MoreRestAPI
FIX - Skype devices should clear caller details when call removed from device
FIX - Internal events relating to Skype objects not correctly removed and causing erroneous log entries
FIX - SkypeTX logging improvements
FIX - Stuck ringing call on SkypeTX codec if call is ringing before the previous one ends
FIX - SkypeTX running connection cycle every 3 seconds even if its correctly connected
FIX - Tieline and Quantum ST codec types missing from configuration list
FIX - PhoneBox server service has to be restarted if SkypeTx server is not avalible when it started or is lost during runtime
FIX - Server FATAL caused by SkypeTX request cancellation
FIX - Server FATAL caused by audio server recording purging
FIX - Uninstall erroneously launching post-install job
FIX - SyncDb password can be removed from server.ini after upgrades
FIX - Question/answer records not reliably being added to new phonecalls

PhoneBOX (General) 3.9.1.41 (STABLE/EAGLE)

Changes since: 3.9.1.35

FIX - Server FATAL caused by SkypeTX request cancellation
FIX - Server FATAL caused by Audio Server recording purging
FIX - Internal events relating to Skype objects not correctly removed and causing erroneous log entries
FIX - Stuck ringing call on SkypeTX codec if call is ringing before the previous one ends
FIX - SkypeTX running connection cycle every 3 seconds even if its correctly connected
FIX - Question/answer records not reliably being added to new phonecalls
FIX - Skype calls disappearing from devices after promotion to video

FIX - SkypeTX logging improvements

PhoneBOX (General) 3.10.0.82 (BETA/FOX)

Changes since: 3.10.0.81

NEW - Add support for websocket keep alives

PhoneBOX (General) 3.10.0.81 (BETA/FOX)

Changes since: 3.10.0.75

FIX - Web configuration issues relating to new x.10.x features
FIX - Skype calls disappearing from devices after promotion to video
FIX - Enable keep alives for sip ws connection.
FIX - SyncDb password can be removed from server.ini after upgrades
FIX - Pathfinder Core routing can stop working after new outputs added to virtual router
FIX - REST API, allow call record tag to be set separately from point

PhoneBOX (General) 3.10.0.75 (BETA/FOX)

Changes since: 3.10.0.74

FIX - 3.10.x.x server installers are much larger than their 3.9.x.x equivalents

PhoneBOX (General) 3.9.1.35 (STABLE/EAGLE)

Changes since: 3.9.1.25

FIX - Comrex codec password failure causes loop of retries draining resources
FIX - UltiDev preqrequisite being downloaded from web unnecessarily
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 - Build to include new client
FIX - E164 / Enhanced number format support for US locale numbers
FIX - VSet caller id appears as sip uri when no name set
FIX - Location lookup broken - Anywhere refactor
FIX - Voip handset not shown if other optional devices are set in the layout but not selected by the user
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 - Rebuild with Sip stack including REFER auth header fix
FIX - Exceptions during answering incoming provider call to sip devices causes stuck calls

PhoneBOX (General) 3.10.0.74 (BETA/FOX)

Changes since: 3.10.0.73

NEW - REST API enhancements - see new documentation

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

PhoneBOX (General) 3.9.1.25 (STABLE/EAGLE)

Changes since: PhoneBOX (General) 3.9.1.24

FIX - Transfer external event not raised while call is on a softphone