Changelog

Changelog

Changelog

2014.09.23 – 3.3.1

  • add added use attended transfer on blf for yealink phones
  • add added fax detection to sip provider settings
  • add added support for Intel 82571/82572/82573/82574/82577/82578/82579/82583 Gigabit Ethernet Controller, and I217/I218 controllers
  • add added t38 option for sip provider
  • update updated Russian translation
  • update updated r8168 driver from 8.031.00 to 8.038.00
  • update updated small fax optimization – 1 second silence for better detection
  • bug fixed infinite loop on wrong did number
  • bug fixed empty catch all pattern within extensions.conf
  • bug fixed sip ddi doesn’t work
  • bug fixed no audio on external phones (inbound SIP, outbound SIP)
  • bug fixed overlap dial is not configured properly
  • bug fixed reload dialplan on ldap configuration
  • bug fixed tiptel 2xx doesn’t work with selected template
  • bug fixed removed persistence db from storage
  • bug fixed ddi header doesn’t work correctly in some cases
  • bug fixed callgroup can’t be changed
  • bug fixed chrash script

2014.09.01 – 3.3

  • add added plug & play support for beroNet cards and gateways
  • add added magic buttons to drop and detect telephony cards
  • add added rpi checkbox to sip provider
  • add added new callerid option for provider: replace callerid name
  • update updated codemirror package to 1.0
  • update updated German translation
  • bug fixed analog phone direct link doesn’t work
  • bug fixed flush boot sequence on every line (it’s easier to detect a freezed function)
  • bug fixed phone detection doesn’t work if no sip phone is configured
  • bug fixed don’t detach fax when initialized
  • bug fixed write fetchmail to syslog
  • bug fixed check if an extension exist when saving a huntgroup
  • bug fixed port groups doesn’t work for DAHDI cards
  • bug fixed gigaset group BLF
  • bug fixed calls doesn’t show up in cdr when routed through callflow
  • bug fixed codecs are missing when an error occured within phone sip edit
  • bug fixed channel language is wrong on local transferred channels
  • bug fixed one way audio on NAT (beronet ISDN/Analog)
  • bug fixed check if extension exist on call groups
  • bug fixed directories for media on storage doesn’t work correctly
  • bug fixed isdn phones doesn’t work within call queues
  • bug fixed wrong IAX/ISDN edit href
  • bug removed restart asterisk every night

2014.07.08 – 3.2

  • add added new phone accounts overview + group edit of phone accounts
  • add changed recommended user number to 100+
  • add added show exact storage size on index page
  • add added show expiry date for test licenses
  • add added stop asterisk gracefully for a reboot
  • add added manual incoming/outgoing/general field in IAX provider
  • add added ‘Show missed calls’ option in queue (CFE)
  • add added enhanced number matching for incoming calls (local phonebook)
  • update updated dropbear package to 2014.63
  • update updated openssh package to 6.6p1
  • update updated stunnel to 5.02
  • update updated msmtp to 1.4.32
  • update updated SQLite to 3080500
  • update updated legal page
  • update updated French translation
  • update updated Gigaset phonebook compatibility
  • update updated German translation
  • bug fixed flush prompts for better debugging on boot
  • bug fixed increased speed of license check
  • bug fixed wrong time on yealink t46g (enabled dhcp time)
  • bug fixed check if SayDigits value is numeric (CFE)
  • bug fixed moved *1 call recording menu to call recording collapsible
  • bug fixed stop asterisk gracefully when restart every night
  • bug fixed don’t send email when email is entered but voicemail to email is deactivated
  • bug fixed if module doesn’t work with less than or greater than (always a string comparison) (CFE)
  • bug fixed accounts/phones is very slow with 50 or more phones
  • bug fixed ext2 storages doesn’t show up as previously used device
  • bug fixed stop monitor even if no recording is made
  • bug fixed force a MWI LED update if voicemail is deleted
  • 2014.06.12 – 3.1.2

    • add added new themes for call control cti
    • add added advanced forwarding for phones on busy or timeout
    • add added enhanced logic for *67 (anonymous calls)
    • add added new checkbox in CFE Extension module – don’t show missed calls on phone
    • add added OA, only_attachments subject codes for virtual outgoing fax (use in email subject)
    • add added c flag to ‘Extension’ module within call flow
    • bug fixed local channels doesn’t contain all internal numbers
    • bug fixed licenses are dopped in some cases
    • bug fixed ISDN on hold problem (no MOH, wrong MOH)
    • bug fixed show reboot message when ‘asterisk restart’ is activated/deactivated
    • bug fixed call recording *1 is also available when global call recording is deactivated
    • bug fixed Call answered elsewhere flag cannot be disabled in localchannels
    • bug fixed empty brackets in provider overview page – if provider doesn’t need register string
    • bug fixed don’t provision phones with logo if ‘Askozia’ logo is deactivated
    • bug fixed use gigaset shortcuts instead of blfs if necessary
    • bug fixed pickup a phone within a callgroup with yealink t46
    • bug fixed web language t46
    • bug fixed disabled dialoginfo callpickup notification
    • bug fixed typos and added translation German

    2014.05.27 – 3.1.1

    • bug fixed CFE/call groups doesn’t work within extension module

    2014.05.22 – 3.1

    • add added call recording for all phone calls
    • add added DND application with BLF
    • add added new checkbox – block all provider by default (when creating a new phone)
    • add added made status logs scrollable
    • add added enhanced status logs
    • add added H264 fmtp profile to video call
    • add added proftpd package
    • add added a reboot message for ‘asterisk restart every night’
    • update updated font color for warning messages on status page
    • update removed storage check on every boot
    • update changed payload type for H264 from 99 to 96
    • update updated German translation
    • update removed weasel application from phonebook by default
    • update updated license strings
    • bug fixed virtual fax issue (missing driver) on embedded systems in 3.0.2
    • bug fixed CDR doesn’t work when unique identifier contains non-alphanumeric strings
    • bug fixed yealink provisioning template
    • bug fixed from name in voicemail representation isn’t saved correctly
    • bug fixed removed hanset volume from polycom 3 firmware template
    • bug fixed callflow numbers can be changed to existing numbers
    • bug fixed don’t send password via url for gigaset and yealink ultra elegant phones
    • bug fixed bug : added busy voicemail (before there was only noanswer)
    • bug fixed ajax.c for better status logs
    • bug fixed updated sounds update procedure
    • bug fixed new providers are not blocked if ‘block all provider when creating a new phone’
    • bug fixed bug in javascript slide routine and some GUI issues

    2014.03.25 – 3.0.2

    • add added port for provider template
    • add added support for multiple provider trunks with unique extensions
    • add added block all provider for all phones with (advanced options -> block all provider)
    • add added sip info only on DTMF on snom phones
    • add added translate context to macro main
    • add added new package jasper for ghostscript
    • update enhanced yealink t4x provisioning
    • update updated translation French
    • update updated translation Russian
    • update updated libtiff to 4.0.3
    • update updated ghostscript to 9.10
    • bug fixed wrong time server settings on yealink phones
    • bug fixed auto answer (intercom) for polycom 4 doesn’t work
    • bug fixed external phone – allow / in dialstring
    • bug fixed notifications email – e-mail me button doesn’t work
    • bug fixed snom provisioning – some buttons are read only
    • bug fixed removed line intend in sip.inc: notifyringing
    • bug fixed ldap request after manual dialplan modifications

    2014.03.04 – 3.0.1

    • add added several embedded performance improvements
    • add added disable voicemail instructions globally (advanced)
    • add added from name field in notification settings
    • add added missing softlinks for yealink_t3x_firmware2.70 template
    • add added automatically archive a queue when too many lines a produced
    • add added garbage collector (resolves some issues with memory leaks)
    • add added 600 seconds as qualify option
    • add added panasonic as phone template (was missing in 3.0)
    • update updated tiptel provisioning to 2x firmware (note drop down list still says 1x)
    • update updated German translation
    • bug fixed stripped fax iso header from subject
    • bug fixed allow forwarding to external phones doesn’t work
    • bug fixed removed backlight from yealink provisioning
    • bug fixed allow to reboot system even without license
    • bug fixed wrong language settings for yealink phones
    • bug fixed automatic phone detection doesn’t work, if no phones with mac address configured
    • bug fixed wrong port on http switcher when https is activated
    • bug fixed block provider within outgoing fax settings won’t be saved
    • bug fixed featuretimout does only work after a restart
    • bug fixed virtual fax – don’t check certificate when no ssl is used
    • bug fixed load allo4xxp driver
    • bug fixed callflow email notification ‘from name’ can’t be changed
    • bug fixed cti protocol is not saved
    • bug fixed don’t play local moh when remote party is putting call on hold
    • bug fixed allow forwarding with 000023 to callgroups and huntgroups
    • bug fixed CTI server produces high load
    • bug fixed https for cti doesn’t work
    • bug fixed wallboard shows ‘submodule’ as extensin number for an incoming call
    • bug fixed set timout on index page statistic to 90 seconds (on embedded systems)
    • bug fixed wrong cti url within admin interface (open cti in a new tab)

    2014.01.20 – 3.0

    • add added fax server (incl. email to fax)
    • add added hot desking
    • add added provisioning for Yealink Ultra Elegant phones (T4X)
    • add added ringtone feature (internal/external)
    • add added new PathSwitch module to CFE
    • add added Caller ID module to CFE
    • add added Caller ID Name module to CFE
    • add added added MusicOnHold module to CFE to change MOH
    • add added conferencing: talk optimization and talk detection
    • add added ext2 as file system for storages
    • add added provisioning for tiptel 31xx
    • add added opvxa24xx driver for generic
    • add added LDAP for phone provisioning
    • add added alternative wallboard design
    • add added new fax archive
    • add added web interface redesign
    • add added new advanced options: allow external forwarding
    • add added pickup dialog info
    • add added scsi support for some controller
    • bug fixed use MSN for calls (multiple phones on one ISDN bus)
    • bug fixed missing restart routine for yealink t3x phones
    • bug fixed sometimes the update failed with: corrupted image
    • bug fixed missing navigation entry for media holiday message upload

    2013.12.19 – 2.2.8

    • add added missing navigation entry for media holiday message upload
    • add added opvxa24xx to generic build
    • add added new ‘use msn’ feature for isdn
    • bug fixed sometimes the update failed with: corrupted image
    • bug fixed wrong label on Queue module
    • bug fixed wrong error message ’0 is not a valid number’ on all special extensions

    2013.10.23 – 2.2.7

    • add added CTI translations
    • add added option ‘qualifyfreq’ to sip phones and providers
    • add added manual dialplan for external phones
    • add added new German translation
    • add added custom audio prompts in webgui
    • add added error message for PressX module in CFE with value 0
    • bug fixed sometimes call view failes with crm_url_push
    • bug fixed cti username is missing in general settings if no cti license is activated
    • bug fixed don’t allow storage formatting if checkbox isn’t checked
    • bug fixed add preferred and asserted identity header after ‘manual dialplan outgoing’ settings in extensions.inc
    • bug fixed only clear sound directory on storage when updating
    • bug fixed call recording for call groups
    • bug fixed create soundfiles directory on storage if it’s missing
    • bug fixed queue statistics in excel
    • bug fixed thank you message also played if when instructions are deactivated
    • bug fixed CDRs are empty on fresh installations
    • bug fixed config is not written after update process
    • bug fixed CTI: outgoing missed call (recent calls shows my id)

    2013.10.01 – 2.2.6

    • add added Call Control CTI module
    • bug fixed NTP reboot problem

    2013.08.14 – 2.2.5

    • add added no more different images for AskoziaPBX and Call Flow Editor
    • add added PRI support for Allo and Openvox cards
    • add added AOC für BRI
    • add added ActionURL module to CFE
    • add added option to restart Askozia every night
    • add added CFE queue option: announce position
    • add added CFE queue option: announce hold time
    • add added the possibility to upload a separated holiday message
    • add added sip qualify selector for sip phones (grandstream workaround)
    • add added added a disabled flag to default provider selector if provider is disabled
    • update updated Russian translation
    • update updated German translation
    • update updated driver for Allo BRI cards (2.6.1)
    • bug fixed call group as pickable extension
    • bug fixed reboot phones after system update
    • bug fixed wrong paging on gigaset phonebook
    • bug fixed gigaset phonebook
    • bug fixed gigaset doesn’t show correct callerid after transfer
    • bug fixed remove empty sip container automatically
    • bug fixed change sip port also in provisioning files if changed
    • bug fixed missing hours in wallboard on active calls
    • bug fixed last added module in CFE is not saved correctly
    • bug fixed tag generation which breaks sip registration for some providers
    • bug fixed phonebook doesn’t work on layer 3 vpn
    • bug fixed voicemail configuration for isdn phones was broken
    • bug fixed snom: user_host doesn’t change within provisioning process
    • bug fixed time doesn’t refresh on startup
    • bug fixed wallboard duration bug (hours weren’t displayed correctly)
    • bug fixed phones doesn’t reboot after AskoziaPBX firmware update
    • bug fixed strange warning messages when deactivating call forwarding via 000023

    2013.05.24 – 2.2.4

    • bug fixed only one entry in phonebook
    • bug fixed Asterisk doesn’t start properly
    • bug fixed remove html specialchars from manual attributes for provisioning templates in phone settings
    • bug fixed empty functionKeys container from snom template if no keys are set
    • bug fixed enhanced safe_asterisk script (run asterisk in the foreground)
    • bug fixed call forwardings to call flows
    • 2013.05.07 – 2.2.3

      • add added Gigaset provisioning for IP 310,410,700 and 900
      • add added the ability to add holidays to nightswitch (thanks @Praktijkcoach)
      • add added magic button for manual file format check
      • add added activate/deactivate outgoing provider contexts for specifics phones
      • add added *67 to block callerID for this/next call
      • add added new checkbox for callgroups: don’t show missed calls on phone
      • add added permanent message if askozia needs a reboot
      • add added hint option for applications
      • add added new CFE template: differentiate between internal and external calls calls
      • add added new option to huntgroups: checkbox to show missed calls on sip phones
      • add added added outgoing callerid map for analog faxes
      • add added the number from dialstring if caller id is not available for agents in wallboard
      • add added new magic button (check for rejected numbers): a quick check for rejected numbers for every provider
      • add added a new toolbar to wallboard
      • add added new call forwarding application 000023*X.*X.
      • add added multiple new settings for notification emails
      • add added show signed in agents to wallboard
      • add added check for sr0 when mounting offload
      • add added option hide last three digits of target callerID in CDR PDF
      • add added possibility to transfer voicemail on call forwarding
      • add added new advanced options in sip phone settings: Intercom – allow auto answer (which adds a new field in provisioning file)
      • add added new advanced feature: disable multiple host handling
      • add added local channels for cfe queues when external phones are used
      • add added yealink 2.7x usernames for phone web interface login
      • add added the ability to change subjects and from fields in notification emails
      • add added the ability to customize the wallboard and cdr logo with usb stick workarround
      • add added dded queue description to wallboard as string
      • add added new feature: set default provider exclusive per phone
      • add added added the possibility to disable the blind transfer fallback: dialplan -> transfers -> miscellaneous
      • add added new tonezone: Saudi Arabia
      • add added new timezone: Pakistan
      • add added function which checks the storage every night for filesystem corruption
      • add added added new kernel option for generic: HyperV support
      • add added mySQL module to CFE
      • update updated snom provisioning: changed some values to read only and active line to writable
      • update updated verification for multiple dial patterns
      • update updated queue call detail records: show call flow name in queue instead of ID
      • update updated German translation
      • update updated notification settings
      • update updated moved misc to the end in dialpplan_transfer.php
      • update updated phone provisioning: check if mac address is already in use
      • update updated asterisk crash script (prevents segfaults on astdb corruption, check if the storage is full if asterisk segfaults)
      • bug fixed voicemail bug which doesn’t work without email address
      • bug fixed bug with wrong fsck.vfat for storage
      • bug fixed bug with X! dialpattern which cause problems on analog faxes, changed it to X. in provider settings
      • bug fixed removed t from fsck.vfat – causes long boot time when usb storage is used
      • bug fixed maxmessage (for maximum voicemail length) to maxsecs (the old variable was deprecated), now 5min
      • bug fixed callgroups -> don’t show missed calls (does work now)
      • bug fixed broken manual pattern selector
      • bug fixed variables in notification settings, fixed missing NOTIFICATION_DATE
      • bug fixed cfe template: changed SayDigits to Text2Speech
      • bug fixed yealink t2x and t3x with 2.7x firmware
      • bug fixed empty page when saving SIP phone
      • bug fixed no keystrokes allowed on phone sip edit page
      • bug fixed bug: cdr entrys don’t have a correct order – sort entrys by timestamp now
      • bug fixed start tftp server at startup if activated
      • bug fixed bug with X! dialpattern which cause problems for analog fax, changed it to X. in provider settings

      2012.12.14 – 2.2.2

      • add added set pai on yealink phones if sendrpid is set on sip phone
      • add added allow one digit numbers in outgoing callerid presentation
      • add added Telbuch as PhCook translation in german translation
      • add added www_provisioning for gettext trranslation
      • add added reboot phones after usb storage restore
      • add added Afghanistan (Kabul) as timezone
      • add added new timzones europe (/usr/share/zoneinfo)
      • add added all Russian timezones
      • add added allow restore if no license is entered (more integrator friendly)
      • add added show name and number on incoming calls on snom phones
      • add added Asia timezones
      • add added MIME header to virtual fax email
      • add added CONFIG_SCSI_AIC79XX to linux config
      • update updated provisioning string for snom phones
      • update updated German translation
      • update updated reboot sip phones before update system
      • update changed panasonic language string from en-UK to en-GB
      • update updated snom provisioning permission flags
      • bug fixed noumea timezone and added noumeo localtime
      • bug fixed push verifysig segfaults to /dev/null
      • bug workarround for start segfault bug
      • bug fixed corrected language path for provisioning files
      • bug smaller design fixes on system_firmware.php
      • bug fixed manual te nt patch for wcb4xx card
      • bug fixed huntgroup members aren’t able to transfer calls

      2012.12.14 – 2.2.2CFE

      • update includes all 2.2.2 changes
      • update removed disabled phones from call flow editor modules
      • update separated contexts for If, IfManually, Switch, SwitchManually and TimeSwitch
      • update updated complex2.json template: Read and Say digits
      • bug fixed recursion bug with wrong extensions on snap back
      • bug fixed cfe bug: no new context for all modules
      • bug fixed bug related to new contexts of if, switch and time switch
      • bug fixed wrong array for chanisavailable output data
      • bug fixed wrong extension in context

      2012.11.09 – 2.2.1

      • add added support for Allo BRI cards
      • add added new network driver: at2711fx
      • add added option ringinuse to queue configuration
      • add added broadcom driver to kernel
      • add added alphabetic order to phone book
      • add added sendrpid and trustrpid to sip phones (advanced options)
      • add added astdb2sqlite3 as asterisk utility
      • add added ssh client
      • add added T.38 gateway capabilities to virtual and analog fax
      • add added new timezone: New Caledonia – Noumea
      • add added new timezone: China
      • add added patch for B800P NT/TE override
      • add added regular expressions to blacklist and greylist fields
      • add added uncheck blacklist anonymous calls if greylist anonymous calls is checked and vice versa
      • add added new ntp server: 0.askozia.pool.ntp.org
      • update updated asterisk to 10.9.0
      • update updated Swedish translation -Thx @ Ulf
      • update updated Danish translation – Thx @ Klaus
      • update updated Frensh translation – Thx @ Rodolphe
      • update updated Russian translation – Thx @ miko.ru
      • update updated transfer to voicemail application
      • update enhanced voicemail menu: you are now able to use voicemail without voicemail to email
      • update merged multiple host fields to one (SIP providers/ advanced options)
      • bug fixed update via GUI routine
      • bug fixed pdf header for correct e-mail attachment
      • bug fixed performance for missed calls on wallboard
      • bug fixed wallboard doesn’t show agents on call if ringinuse is set to no
      • bug fixed call recording header bug
      • bug fixed extension instead of admin for Yealink login
      • bug smaller gui fix in navigation
      • bug virtual fax: removed uuencode in favor of base64 for correct attachments
      • bug changed wav format from wav49 to normal wav: caused transcoding bug in app_voicemail
      • bug fixed broken incoming extension selector in French
      • bug exclude some languages from CDRs (special characters issues)
      • bug fixed storage needed for voicemail to email
      • bug fixed reboot phones after firmware update

      2012.11.09 – 2.2.1CFE

      • update includes all 2.2.1 changes
      • add added leavewhenempty to queue module
      • add added joinempty to queue module
      • add added GoTo after Timeout-Extension possible
      • add added GoTo after Invalid-Extension possible
      • add added Press Extension after Timeout-Extension possible
      • add added GoTo to extension 1-9 possible
      • bug fixed switch and if manually modules

      2012.08.15 – 2.2CFE

      • update includes all 2.2 changes
      • add added wallboard (live call statistics)
      • add added agent login for queues
      • add added queue statistics (.csv) in CDR menu
      • add added duplicate call flows
      • add added manual if and switch modules (define your own variables or use any Asterisk variable)
      • add added ringing instead MOH (queue module)
      • add GUI design updated
      • add added new options to email module (add date, callerid and duration)
      • add added ability to change font size in wallboard
      • add added statistic user login for wallboard (general settings)
      • add added German translation for CFE
      • bug fixed SendDTMF module

      2012.08.15 – 2.2

      • add added call recording
      • add added automatic phone detection
      • add added TFTP
      • add added backup external storage
      • add added auto firmware update via GUI
      • add added black and grey listing
      • add added provisioning for Alcatel IP phones
      • add added CDRs to .csv
      • add added astman proxy
      • add added new enhanced logging message viewer
      • add added added Polycom 4.x firmware provisioning support
      • add added provider name to call notifications
      • add added provisioning for Huawei phones
      • add added missing voicemailbox feature for skinny (beta)
      • add added t38 gateway support
      • add added new driver: Realtek r8168 gigabit network driver
      • update updated to Asterisk 10 and DAHDI 2.6.1
      • update updated German translation (thanks @ Raiko)
      • update landing page
      • update updated to Linux kernel 2.6.39.2
      • update removed redfone gateway from navigation
      • update removed intelligent call back
      • update license management
      • update changed ‘Media’ string to ‘Media upload’
      • bug fixed CSS bug in web interface
      • bug fixed configuration upload does not work
      • bug fixed nightswitch media upload does not appear in navigation

      2012.08.15 – 2.1.7CFE

      • update includes all 2.17 changes
      • bug fixed call answered elsewhere (no missed calls if call was answered by a member of the group)
      • bug fixed special characters result in exiting call flow
      • bug allow capital letters in filetypes
      • bug improved description for If and Switch modules
      • bug make arrow visible if the next module is behind the module before
      • bug fixed goto doesn’t work anymore if order of modules was changed
      • bug fixed various scrolling problems
      • bug fixed AVAILSTATUS in ChanIsAvailable module
      • bug fixed positions in complex templates
      • bug fixed a few strings in call flow modules
      • bug fixed call forwarding doesn’t work – caused by php header in shell
      • bug added default values to busy and congestion modules
      • bug show error message of module for goto is not found
      • bug fixed fixed upload button in safari
      • bug removed double module entries
      • bug fixed fixed internet explorer 8.0 upload bug
      • bug added callflows to phonebook
      • bug fixed updated yui connection library from 2.6.0 to 2.9.0 (fixed bug: safari record button doesn’t work)
      • bug fixed updated ChanIsAvail template
      • bug fixed large call flows import problem
      • bug fixed modules out of reach in some cases

      2012.08.15 – 2.1.7

      • add added reboot all phones before rebooting
      • add added new CDR filter: exclude outgoing or incoming calls
      • add added phonebook and nightswitch exceptions for applications
      • add added snom 80x auto provisioning
      • add added new dialog: please make a backup before updating
      • add added added ISDN transfer capabilitys
      • add added blind transfer fallback
      • add added some enhancements for incoming calls: rejected to voicemail, wait one second before conferencing
      • add added wait two seconds after failed blind transfer
      • add added npi+ton selector for ISDN providers
      • add added c option to Dial and Macro (main in callgroups and huntgroups)
      • add added check for similar dial pattern before saving provider settings
      • add added allow to enter multiple hosts for multiple contexts if a provider provides multiple SRV records
      • add added busy detection, polarityswitch option to analog phones and faxes
      • add added ISDN to DAHDI tonezone description
      • add added added new prov.php for yealink t3x phones
      • update updated French translation
      • update updated humbug to latest release
      • update updated provisioning: disabled auto update for snom phones
      • update updated Danish translations
      • bug fixed USB boot fails
      • bug fixed nightswitch overlap bug
      • bug fixed semicolon in characters break passwords
      • bug fixed special character breaks phone edit link
      • bug fixed list for nightswitch forwarding
      • bug fixed humbug-collector start routine
      • bug fixed directed pickup for groups/queues and analog phones
      • bug fixed nightswitch media upload does not appear in navigation
      • bug fixed page format doesn’t work for CDRs

      2012.04.26 – 2.1.6CFE

      • add added Dial Number module
      • add added Time Switch module
      • add added new templates
      • add added all extensions to Queue Member module
      • add added Email module
      • update updated French translation; Many thanks @ Rodolphe from ordiservices.ch
      • bug fixed no linebreaks in text boxes
      • bug fixed night switch bug – sun/mon overlap
      • bug fixed wrong postioning after reload
      • update all other 2.1.6 changes included

      2012.03.09 – 2.1.6

      • add added magic buttons in integrator panel (e.g. make unused space on CF available)
      • update updated date presentation for Askozia mail system
      • bug fixed different error and notify messages
      • bug fixed refresh gui css after update from 2.0
      • bug fixed nightswitch bug (days shifted)
      • bug fixed nightswitch bug caused by translation
      • bug fixed Polish voicemail
      • bug fixed DNS timeouts – added option to disable provider templates
      • bug fixed chaning ISDN options makes Warp crash
      • bug fixed CDR presentation bug (overlapped strings)

      2011.12.23 – 2.1.5

      • add added hunt groups
      • add added error messages for missing storage
      • add added pipe as possible value for snom phones (101|*8 for one-touch pickup)
      • update updated French translation
      • update updated German translation
      • bug fixed no internet connection results in a slow web interface
      • bug fixed picking up an extension with only one digit fails
      • bug fixed embedded issue with B100M OpenVox cards
      • bug fixed fax to e-mail doesn’t work in some cases
      • bug fixed night switch message chopped at the beginning (ISDN)
      • bug fixed don’t generate coredumps if Asterisk crashes
      • bug fixed drag-and-drop when all de-selected
      • bug fixed phone book for snom phones
      • bug fixed virtual fax doesn’t work when night switch is activated
      • bug fixed not more than 6 speed dial buttons
      • bug fixed failover provider
      • bug fixed some error messages in log regarding dialplan
      • bug fixed wrong callerid prevent Snom auto-configuration
      • bug fixed night switch week days switch automatically to next day after saving
      • bug fixed changing from automated to manual night switch results in a lost of time data
      • bug fixed overlap digits isn’t checked after saving

      2011.12.02 – 2.1.4

      • add auto-configuration of Yealink, Tiptel, Snom, Panasonic, Sipura and Linksys
      • add provider templates
      • add night switch
      • add support button
      • add global phone book
      • add speed dial button configuration
      • add preview for caller ID
      • add activation key can survive a factory reset
      • add euroisdn is now selectable in GUI
      • add warning in GUI if config files were changed
      • add SIP permit/deny mask
      • add ISDN lines now also have a status light in GUI
      • add direct link to web interface of VoIP phones
      • add time zones Venezuela and Mosambique
      • add intelligent callback for SIP and analog
      • add “+” in dialstring is replaced by “00″
      • update forwarding to external phone
      • update file names of uploaded files (e.g. MOH) are now in GUI
      • update outgoing patterns simplified
      • bug virtual faxes now appear in CDRs
      • bug overlapdial for ISDN is now default
      • bug no group restarts of auto-configured phones
      • bug busy level fixed

      2011.07.29 – 2.1.3

      • add added From User can be disabled
      • update updated Asterisk to 1.8.4.4
      • update included Asterisk Cisco patch
      • bug fixed dahdi.inc error message
      • bug fixed Safari 5.1 integrator panel issue
      • bug fixed voicemail doesn’t work after 2.0 update

      internal release – 2.1.2

      • bug fixed readbacknumber now works for incoming calls
      • bug fixed call forwarding transfer bug
      • bug fixed format button is never “active”
      • bug fixed some smaller security issues

      2011.07.07 – 2.1.1

      • add added option “originate” in the manager interface
      • add added default date for CDR generation
      • update updated Polish translation
      • update updated CDR information for analog lines
      • bug fixed pickup (*8) results in Asterisk crash
      • bug fixed standard voicemail
      • bug fixed different error and notify messages
      • bug fixed logout button for HTTPS
      • bug fixed “unable to park a call twice”
      • bug fixed option “outgoing calls only” can not be selected
      • bug fixed special characters in SIP registration not supported
      • bug fixed application “ME” is available after update
      • bug fixed SIP phones do not hang up after parking a call
      • bug fixed incoming calls result in log endless loop
      • bug fixed incoming calls coming in through analog do not appear in the CDRs

      2011.06.24 – 2.1

      • add added mp3 support for music-on-hold
      • add added new web interface design
      • add added SNOM auto provisioning
      • add added virtual fax
      • add added fax archive
      • add added automatic fax detection
      • add added fax to e-mail
      • add added voicemail system
      • add added call forwarding
      • add added call waiting
      • add added call forwarding
      • add added pickup groups
      • add added call detail records to PDF
      • add added humbug fraud detection
      • update updated German translation
      • update updated Asterisk to 1.8.4
      • update updated jquery to 1.8.13
      • update Blackfin CPUs are no longer supported

      2011.04.11 – 2.0.4

      • add added custom music-on-hold
      • add added application ME(000063)
      • add added restore function to storage services
      • add added analog fax
      • add added automatic update check
      • add added hardware graphic to summary
      • add added sox for media file conversion
      • add added custom voicemail greetings
      • add added Humbug fraud detection (beta)
      • add added favicon
      • add added DHCP fallback
      • update updated Danish translation
      • update updated Italian translation
      • update updated Polish translation
      • update updated German translation
      • update updated Asterisk to 1.6.2.17
      • bug fixed changing the password terminates PHP process
      • bug fixed IE navigation bar issue
      • bug fixed a couple of small bugs

      2010.11.26 – 2.0.3

      • add ISDN Phone ports and accounts now auto-configured on boot
      • update updated German translation
      • refactor removed translation and hardware tabs from integrator panel, improved gui loading speed
      • bug more blind and attended transfer issues resolved
      • bug missed call notifications now being sent in all cases
      • bug firmware image size incompatibilities when using physdiskwrite with certain card readers on Windows resolved
      • bug fixed IP information in message when resetting to factory defaults
      • bug echo cancellation is now off by default on ISDN ports to work around CPU load issues on certain platforms
      • notice: ISDN issues remain
      • notice: this release requires more than 32MB of storage, 64MB CF cards should be used

      2010.10.22 – 2.0.2

      • add added Portuguese (Brazil) voice prompts
      • add ISDN Providers now have options for international, national, local and private prefixes
      • add Providers now have the option of prefixing digits to incoming and outgoing caller ids
      • add added support for the OpenVox A800 and A1200 cards
      • update updated Spanish translation
      • update updated Dutch translation
      • update updated Asterisk to 1.6.1.20
      • bug blind and attended transfer issues resolved
      • notice: this release requires more than 32MB of storage, 64MB CF cards should be used

      2010.06.03 – 2.0.1

      • add added Russian voice prompts (by ivrvoice.ru)
      • add translation percentages now displayed next to incomplete webgui languages
      • add port status page now provides minimal information about isdn and analog ports
      • update updated German translation
      • update updated Danish translation
      • update updated Polish translation
      • update updated Russian translation
      • bug fixed bug when changing the administrator username
      • bug option to add new isdn phone account only present if an appropriate port is available
      • bug reboots not required as often after General Setup page changes
      • bug fixed outgoing calls to Analog providers after reboots
      • bug fixed strange non-english prompt behavior

      2010.05.04 – 2.0.0

      • add added Dutch voice prompts (by borndigital.nl)
      • add added Polish voice prompts (by Mateusz Viste)
      • add added ISDN Phone support
      • add added Q-Stat System Performance Statistics recording page (available in the beta features tab of the Integrator Panel)
      • add added option to disable e-mail server certificate checking for servers not yet recognized by AskoziaPBX
      • add added sftp support (by Devon Hendricks)
      • update updated Bulgarian translation
      • update updated Dutch translation
      • update updated French translation
      • update updated German translation
      • update updated Italian translation
      • update updated Polish translation
      • bug music-on-hold configuration now editable and working correctly when media storage service is activated
      • bug fixed ssh path settings (by Mats Karlsson)
      • bug fixed “from” header on missed call notification e-mails
      • bug sshd keys now generated on a per-install basis (by Devon Hendricks)
      • bug disabled https option and Snom auto-provisioning feature temporarily until they are production ready

      2010.04.09 – 2.0.rc3

      • add added IAX2 RSA key authentication options to IAX2 Providers (sponsored by Steve Gray of Data Distribution Systems / Global Audio Video)
      • add added regional analog compatibility options (by Giovanni Vallesi)
      • add added Storage Disk interface
      • add added Media storage service for external storage of voice prompts and music-on-hold
      • add added Media Manager to install additional voice prompt packages
      • add added Persistence storage service for external storage of Asterisk’s database
      • add added SMTPS support to E-Mail Notifications (by Georg)
      • add added support for ISA network cards
      • add added support for USB keyboards
      • bug External Phones no longer forget their dialing provider
      • bug provisioning services now only started when needed
      • bug fixed WakeMe wake up call application (by Devon Hendricks)
      • bug fixed BLF behavior for SIP Phone accounts
      • bug fixed issue with Provider Accounts’ generated settings in chan_dahdi.conf
      • bug fixed timezone settings in Asterisk
      • update updated Asterisk to 1.6.1.18
      • update updated MSMTP to 1.4.19
      • update updated Busybox to 1.15.3 (by Devon Hendricks)
      • bug COMpact 3000 : led control and system initialization improvements (by Jens Möller)

      2010.03.12 – 2.0.rc2

      • add added beta rework of Dialplan Applications: PHP, syntax highlighting, Flite text-to-speech engine (press ESC, click on beta features)
      • add added Phone and Provider connectivity status bubbles to Accounts overview page
      • add added Provider Port Grouping to Analog and ISDN Telephony Ports
      • add added support for many single-port Analog and single-port ISDN cards
      • add added a Dialplan Application which reads back the system IP (0000IP)
      • add added support for SATA controllers
      • add many new countries indication tones supported
      • add increased support for multiple-port ISDN cards
      • add increased support for IDE controllers
      • add upgraded echo canceller to use OSLEC
      • add more robust analog module detection (by Giovanni Vallesi)
      • add more robust detection of Live CDs
      • bug fixed display of user defined Voicemail E-Mail text
      • bug fixed Application Element Library’s prompt file display
      • bug fixed missed-call notifications
      • bug fixed external phones which use Analog or ISDN Providers
      • bug fixed call groups behavior
      • bug fixed analog hardware port selection in Phones and Providers
      • bug fixed generation of ISDN configuration, multiple b-channels now usable
      • bug fixed default channel language in applications (by Devon Hendricks)
      • bug fixed WakeMe (time fix still needed) (by Devon Hendricks)
      • bug fixed filename truncation on Storage Disks (press ESC, click on beta features)
      • bug serial console settings fixed (by Stephane Billiart)
      • update updated Asterisk to 1.6.1.17
      • add COMpact 3000 : Analog ports now supported
      • bug COMpact 3000 : ISDN configuration generation fixed
      • bug COMpact 3000 : upgrading firmware via the WebGUI now much safer

      2010.02.13 – 2.0.rc1

      • add Moved from FreeBSD to branch of T2 Linux
      • add Moved from Asterisk® 1.4 to 1.6.1
      • add Many pages completely rewritten with lightweight GUI input and validation framework
      • add Auto-detection and configuration of Analog ports and Phone accounts
      • add Auto-detection and configuration of ISDN ports
      • add Auto-provisioning of Snom telephones
      • add Configuration of Redfone gateways
      • add Blackfin CPU architecture support
      • add Manual configuration changes supported
      • add External storage device support
      • add Provider failover support
      • add Skinny telephone support
      • add Integrator Panel (press ‘ESC’ in WebGUI to activate)
      • add …plus innumerable smaller changes

      1.0 Series


      2009.05.29 – 1.0.3

      • add French localization
      • add Japanese voicemail template
      • update updated German localization

      2009.03.10 – 1.0.2

      • add Turkish localization
      • add Japanese localization
      • add Spanish localization
      • update updated Bulgarian and Dutch localizations
      • refactor pbxdev.php extended to allow for both freebsd and linux to be used as the base operating system
      • refactor navigation menu changed to use <ul> and <li> elements

      2008.09.19 – 1.0.1

      • bug providers with non-numeric usernames with no “read back” number set no longer crash in voicemail
      • bug deleted phone accounts now automatically removed from call groups
      • bug call groups with no members no longer break dialplan
      • bug busy/call limit documentation fixed
      • bug the “invalid input” state of SIP/IAX provider/phone account pages no longer results in the selected codecs being reset to defaults
      • bug Greek translation is now in Greek instead of Bulgarian
      • bug sqlite CDRs now store year information

      2008.09.12 – 1.0.0

      • add Czech localization
      • update updated German, Bulgarian and Italian localizations
      • changed CD now carries over config changes upon installation
      • changed CD installs directly from medium instead of memory disk, saving memory
      • bug disk name extraction fixed
      • bug custom voicemail subject lines with quotes in them now saved/displayed correctly (reported by devon in the forums)
      • refactor sounds reorganized for easier package creation

      2008.08.21 – pb14.3

      • add outgoing Caller ID options added to ISDN and Analog Providers
      • add Portuguese (Brazil) audio prompts
      • add Greek localization
      • update updated German, Italian and Dutch localizations
      • update scriptaculous to 1.8.1
      • bug Systems with Cyrix 5530 ATA controllers now working
      • bug ACPI issues fixed, Intel D201GL* boards now working
      • bug “Remote UNIX connection” messages no longer generated
      • bug Caller ID and Caller ID String fields now verified
      • refactor jQuery cleanups

      2008.08.15 – pb14.2

      • add SMTP settings can now be tested via “Services -> Voicemail”
      • add Polish localization
      • add German localization
      • add Bulgarian localization
      • update PHP to 4.4.9
      • bug reverting to jquery 1.2.1 – fixes a few javascript incompatibilities introduced in 1.2.6
      • refactor major gettext cleanups / fixes

      2008.08.08 – pb14.1

      • add log reverse sort order option reinstated
      • add SIP & IAX2 URIs are now usable as dialstrings for external phone accounts
      • add user definable voicemail notification e-mail text
      • add Simplified and Traditional Chinese localizations
      • add Dutch localization
      • add Danish localization
      • update jquery to 1.2.6
      • bug zoneedit dynamic dns update server address corrected

      2008.07.31 – pb14

      • new platform Live + Install CD
      • add system storage media larger than 96MB will automatically be partitioned with a permanent storage partition
      • add basic package management system and api with backup, restore, activate, deactivate and delete
      • add providers and phones may now be disabled / enabled
      • add SIP and IAX providers and phones now have icon indicating connection status
      • add webGUI gettext localization
      • add Finnish and Italian language translations + skeleton files for en, es, da, de, fr, it, pl, nl, pt, fi, se and ru
      • add a more secure, machine-specific HTTPS certificate is generated on the first boot if not defined
      • add log display page pagination + filtering
      • add appropriate interfaces are now checked for before ISDN / Analog accounts can be set up
      • add if default or configured network interfaces are not detected, new working settings are generated
      • add dynamic DNS update support
      • add page specific help has been added on pages bearing the “?” icon next to their title
      • add logging package which enables permanent storage of system, pbx and call logs
      • update Asterisk 1.4.21.2
      • update isdn4bsd r751
      • update FreeBSD 6.3-RELEASE-p2
      • “Accounts -> Providers / Phones” pages redesigned
      • reworded help text on many features
      • seldom used ISDN interface settings moved to “advanced” settings, timing defaults improved
      • seldom used Analog interface settings moved to “advanced” settings
      • isdn4bsd and generic usb devices are now compiled as modules
      • bug fixed “help text” display bug on ISDN and Analog interface summary pages
      • bug fixed music-on-hold for ISDN accounts
      • removed log reverse sort order option remove

      2008.06.05 – pb13.4

      • bug using applications as a Provider’s incoming destination works in more cases
      • bug incoming calls from SIP Providers are now accepted in more cases

      2008.03.27 – pb13.3

      • add isdn and analog interface settings can now be “forgotten” in the webGUI
      • add dialplan now produces more human-readable log messages
      • add Page() application to base distribution
      • add manual attributes can now be defined for ISDN interfaces
      • add “readback” numbers (used for unreachable messages) can now be set manually for SIP and IAX providers
      • add Voicemail enabled extensions now have an option to signal “busy” via tones instead of going to Voicemail
      • add LAN DNS IP now configurable via console (patch provided by devon in the forums, small fix needed)
      • busy extensions are signaled via tones for extensions without Voicemail enabled
      • internal unique ids are no longer converted to names on the “Diagnostics -> Logs -> PBX” page
      • bug incoming calls from ISDN providers will now be accepted in more cases
      • bug ISDN and Analog interfaces are now automatically renamed from “(unconfigured)” upon configuration
      • bug ISDN Operating Mode is now verified before saving
      • bug missed call notifications are no longer sent when a voicemail message was left
      • bug using applications as a Provider’s incoming destination works properly again (potentially still not working) (reported by Marco in the forums)
      • refactor outgoing SIP uri dialing logic has been simplified
      • refactor main macro completely rewritten

      2008.03.12 – pb13.2

      • add transmit and receive gains can now be set for analog interfaces (working patch provided by devon in the forums, modified for code consistency)
      • add improved documentation on the analog and isdn interfaces pages
      • add manual attributes can now be defined for analog interfaces
      • add an authentication method can now be selected for the SMTP server used in “Services -> Voicemail”
      • * and # characters may now be used in application extensions
      • bug missed call notifications are no longer sent for successfully completed calls
      • bug incoming calls from SIP or IAX providers landing in voicemail will now be read back the account’s username if it is numeric instead of the internal extension
      • bug incoming calls from providers will now be accepted in more cases (previously only numeric and ’s’ extensions would be matched, now all extensions containing alphanumeric, # and * characters will be matched)
      • removed echo squelch options removed from isdn interfaces as it is no longer supported in isdn4bsd

      2008.02.26 – pb13.1

      • add incoming caller id from providers may be prepended or replaced by a user defined string
      • add NAT settings can now be overridden under “Advanced Settings” in SIP accounts
      • add phones and callgroups now have selectable ring lengths
      • SIP phone accounts now always have nat=yes set
      • bug multiple SIP Provider accounts on the same host are now correctly routed (working *.conf example provided by Sergio in the forums)
      • bug boot messages are no longer being suppressed by a poorly chosen variable name in extensions.inc
      • bug outgoing caller id overrides in providers are now functional
      • bug diag_editor.php no longer inserts unwanted ‘\r’ characters and automatically remounts /conf if needed (reported and patched by devon in the forums)
      • bug Applications are no longer generated with ’s’ extensions, rather ‘X!’ patterns. This allows the application to be aware of which extension it was reached with. (reported by ciscomonkey in the forums)

      2008.02.14 – pb13

      • add manual attributes may now be defined for phones, providers and under “Advanced” for SIP and IAX technologies
      • add custom application logic may now be defined in “Dialplan -> Applications” (suggested w/proof-of-concept by Ben Hathaway)
      • add factory default reset button support for alix23x platform (merged from m0n0wall)
      • add Russian voicemail notification e-mail translation (submitted by Eugen Bernatskiy)
      • add Portuguese voicemail notification e-mail translation (submitted by Marcus Vinícius Quintella Ribeiro)
      • add “DNS Service Records” option to “Advanced -> SIP” so SRV lookups can be disabled
      • add ‘!’ characters are now allowed in incoming and outgoing dialpatterns
      • add provider dialpatterns now allow ‘#’ and ‘*’ characters (suggested by Dave Fear)
      • add software package versions used in each release now listed in /etc/versions
      • add when defining the incoming routing for a provider, impromptu call groups can now be set up by defining two destinations with the same pattern
      • add phones, callgroups, conferences and applications have a new option “Public Direct Dial” which, when activated, exposes these extensions to public networks. An optional string may also be defined to override the internal extension with a friendly name (“yourname” vs “1234″)
      • add ajax.cgi allows execution of Asterisk Manager Interface and shell commands
      • add jQuery plugin copyright information to license page
      • add direct outgoing sip uri dialing (unfinished: cid options…)
      • add more information in printable dialplan
      • add Danish language audio prompts (GSM) and voicemail notification e-mail translation (provided by McM in the forums)
      • update Asterisk to 1.4.17
      • update php to 4.4.8
      • update core sounds to 1.4.8
      • update extra sounds to 1.4.7
      • update basesystem to FreeBSD 6.2-RELEASE-p10
      • update timezone information (merged from m0n0wall)
      • additional sounds used for WakeMe application are now in the higher quality ulaw format
      • callgroup member selector now displays the phone’s extension
      • “Diagnostics -> Manager Interface” now uses AJAX to query the new ajax.cgi backend
      • /exec.php now uses AJAX to query the new ajax.cgi backend
      • existing GUI javascript code replaced with jQuery equivalents where possible
      • bug fixed bridging with interfaces that support hardware TX checksumming
        (by turning it off) (merged from m0n0wall)
      • bug added patches to fix rebooting on alix boards (merged from m0n0wall)
      • bug added patches to fix trap 12 kernel panics on Nokia IP110/IP120/IP130 (merged from m0n0wall)
      • bug call records are now sorted properly (reported by Jakob Strebel)
      • bug call groups now define their “read back” number properly (reported by Janåke Rönnblom)
      • bug external phones reachable via ISDN providers working properly again
      • bug nge network interfaces are no longer ignored (merged from m0n0wall)
      • bug omit no-cache headers on exec.php because it confuses IE with file downloads (merged from m0n0wall)
      • bug “Services -> Voicemail” now properly sets the serveremail property in voicemail.conf (reported by Falko Mach)
      • bug added missing newlines to iax.conf generator (reported by Falko Mach)
      • bug dtmf tones are no longer played after picking up a ringing Analog phone (fixed by David G. Lawrence)
      • bug extensions are now gathered properly from phones (reported by devon in the forums)
      • refactor all providers and phones moved to subsections under “Accounts”
      • refactor individual sorting functions replaced with pbx_sort_by_xxx() functions
      • refactor all asterisk_* functions renamed to pbx_*
      • refactor all manager, rtp and indications related functions moved from pbx.inc to manager.inc, rtp.inc and indications.inc
      • refactor all features, application and callgroup related functions moved from dialplan.inc to features.inc, applications.inc and callgroups.inc
      • refactor all verification and network related functions moved from util.inc to verify.inc and network.inc
      • refactor added isdn_get_provider()
      • refactor extension generator is much cleaner now

      2007.11.16 – pb12.2

      • bug deleting a phone or callgroup no longer ends up in a page hang (reported by Mattijs V)
      • bug outgoing calls are no longer limited to a 20 second ring time (reported by Andreas J)
      • bug ISDN phones and providers now have their prompt language set properly (reported by Kai D)

      2007.11.09 – pb12.1

      • bug cleans up errors left by config.xml upgrade bug in pb12 (reported by Carlo L)

      2007.11.09 – pb12

      • new platform PC Engines – ALIX.2,3x
      • add outgoing caller id override options to SIP and IAX providers
      • add English (UK) ulaw prompts, renamed existing prompts to English (USA)
      • add French (France) gsm prompts, renamed existing prompts to French (Canada)
      • add missed call notification e-mail option to phones
      • add chan_local jitterbuffer patch, enabling applications to also be jitterbuffered
      • add hints to call parking spots based on info provided by Mat M
      • add multiple incoming extensions per provider can now be defined
      • add multiple, individually addressable, ISDN phones may now be connected to a single port
      • update Asterisk 1.4.13
      • cleaned up iax.conf generator based on suggestions by Mat M
      • replaced internal call detail record logging with sqlite backend
      • changed default Dial timeout to 20 seconds, still needs to be made a configurable option
      • bug fixed “localnet” setting in sip.conf
      • bug fixed snooze feature in WakeMe
      • bug “Diagnostics -> Logs -> Calls” now displays information in the “src” field in more cases
      • bug fixed a small display issue when a provider does not have any patterns defined
      • bug unconfigured analog interfaces now have a default value of “128″ for their echo cancellers instead of “yes”
      • removed removed gsm prompts from languages which also have ulaw prompts to make room for more languages
      • removed disabled “Dialplan -> Providers” page as it is currently broken, several things should be rewritten before yet another workaround is implemented

      2007.10.09 – pb11.1

      • add missing OSLEC information on license page
      • update Asterisk 1.4.12.1
      • bug unwanted wireless information no longer displayed on systems with no wireless interfaces present (reported by Carlo L)
      • bug manager is now bound to 0.0.0.0, allowing connections from users defined in “Advanced -> Manager” (reported by Carlo L)
      • bug added forgotten “-incoming” to iax.conf generator (reported by Mat M)

      2007.10.05 – pb11

      • new platform PC Engines – ALIX.1x
      • new platform Soekris – net55xx
      • new platform Herologic – HL-4xx
      • add “start signaling” option to Analog interfaces
      • add “echo cancel” option to Analog interfaces, ported OSLEC (http://rowetel.com/ucasterisk/oslec) to FreeBSD for this and some testing is still needed
      • add “Dialplan -> Applications” page so applications may be mapped to the dialplan
      • add Wireless interface support
      • add “Advanced -> Manager” page to allow extra AMI users to be defined
      • add additional network interfaces may now be bridged to the “main” interface
      • add resetting to factory defaults now sets values appropriate for each platform
      • update Asterisk 1.4.12
      • update script.aculo.us 1.7.1 beta 3
      • improved incoming extension matching
      • “Interfaces” menu collapsed into a single tabbed page under “System -> Interfaces”
      • bug removed channel queue limit patch which was dropping frames on slower hardware until a tunable parameter can be implemented
      • bug fixed invalid options being stored in incoming extension
      • bug fixed extension generator for callgroups not having an internal extension defined
      • bug fixed NAT configuration generator in sip.conf
      • bug fixed dialplan_parse_pattern()

      2007.09.07 – pb10

      • add Polish voicemail notification e-mail translation
      • add Russian language (gsm) audio prompts
      • add MAC spoofing support for the network interface
      • add “attended transfer answer,” “transfer key” and “extension digit” timeout options to “Dialplan -> Transfers”
      • add collapsable “Advanced” menu and option to keep it open in “Advanced -> GUI Options”
      • add options in “Advanced -> GUI Options” to hide menu entries for unused telephony technologies
      • add call-limit and busy-limit options to SIP Phones and fixed SIP hints in dialplan, basic presence information is now working
      • add WakeMe – Wake-Up Call Manager to extension 00009253 (0000WAKE)
      • add a basic printable Dialplan
      • add “Advanced -> RTP” page so the RTP port range can be defined
      • add initial Analog Phone support
      • add debugging tools for USB ISDN devices
      • add callers from isdn and analog providers are now read the external telephone number and not the internal extension when reaching a voicemail account or timing out
      • update Asterisk 1.4.11
      • update FreeBSD 6.2-RELEASE-p7
      • improved device detection for Analog FXO modules / cards
      • improved the default Dial() flags: transfer permissions and caller id strings should “make sense” more often
      • bug fixed the broken jitterbuffer options on in “Advanced -> IAX”
      • bug fixed sound issues introduced in pb9 in certain situations (IAX channels are still having some problems)
      • bug incoming extension reference deletion is now centralized and used in previously omitted cases
      • bug configured, but absent analog interfaces are no longer initialized (thus failing) during boot and during Asterisk restarts
      • bug ab-units are no longer displayed on “Interfaces -> Network -> Assign”

      2007.08.09 – pb9

      • add multilingual voicemail notification e-mail option (en, de, it, nl, fr, es, se)
      • add “Indications Tonezone” selector to the “System -> General Setup” page
      • add Analog Interface and Provider support (very basic)
      • add only appropriate interfaces are displayed when adding an ISDN/Analog Provider/Phone
      • update Asterisk 1.4.10
      • bug incoming calls from IAX providers should pass in more cases now

      2007.08.02 – pb8

      • add ISDN Phone support
      • add External Phone support (phones not directly connected to but accessible from AskoziaPBX)
      • add Jitterbuffer enable and force options to the “Advanced -> IAX” page
      • add md5 authentication option to IAX accounts
      • add ISDN interface information to “Status -> Interfaces” page
      • add transfer key combination options and changed default combinations to:
        • attended transfer = “**”
        • blind transfer = “##”
      • add call groups can now be mapped to an internal extension
      • add removed the 4 digit limitation to internal phone numbers
      • renamed “Dialplan -> Call Parking” page to “Transfers”
      • bug conference delay and moh issues fixed by code submitted to the asterisk-bsd list by David G. Lawrence
      • bug fixed unchecked array in asterisk_dialpattern_exists()
      • bug USB ISDN cards attached after AskoziaPBX has already booted are now detected
      • bug incoming calls from IAX providers are now handled better

      2007.07.26 – pb7

      • removed platform removed net45xx platform support until test hardware can be acquired
      • add verbosity, internal_timing and highpriority now set in asterisk.conf
      • add logs now display friendly names instead of internal unique ids
      • add incoming Caller ID name from providers may now be overridden with the incoming Caller ID number
      • add “Status -> Channels” page which displays all currently active channels
      • add “Status -> Conferences” page which displays all currently active conferences and allows members to be kicked from conferences
      • add “Dialplan -> Call Parking” page to manage parking extensions
      • add ISDN Interface and Provider support
      • update Asterisk 1.4.9
      • moved “assign” link into “Interfaces -> Network” page as a tab
      • cleaned up “Manager Interface” output
      • bug disabled core dumping of Asterisk
      • bug calling an unregistered user with no voicemail account now results in a message instead of an abrupt hangup
      • bug fixed handling of IAX providers with no patterns set
      • bug fixed missing IAX ’s’ extension
      • bug incoming extension references to call groups are now removed upon call group deletion
      • bug numbers now sort like numbers
      • bug disabled core dumping of Asterisk

      2007.07.05 – pb6

      • add improved sound packaging to finally include only what is necessary
      • add dialplan now has an ’s’ extension to pick up stray incoming calls from providers, this is helpful with at least one ATA device until a better solution is implemented
      • add help messages now only appear once per page when multiple instances of the same field are present
      • add phones and providers appear properly sorted now (earlier first appeared sorted SIP then sorted IAX)
      • add multiple patterns may now be entered
      • add audio prompts for the following languages:
        • Dutch (gsm)
        • French (gsm/ulaw)
        • German (gsm)
        • Italian (gsm)
        • Japanese (gsm)
        • Spanish (gsm/ulaw)
        • Swedish (gsm)
      • add registration timeout options to sip providers
      • add parallel ringing call group support
      • add iLBC and Speex codecs
      • update FreeBSD 6.2-RELEASE-p5
      • update Asterisk to 1.4.6
      • normalized some pages’ POST and redirection routines
      • removed removed prefixes (automatically converted to patterns on the first boot)

      2007.06.21 – pb5

      • add “qualify” options to sip/iax phones/providers
      • add pattern matching support to sip/iax providers
      • add prefix/pattern setting to “Dialplan -> Providers” page
      • add active call/channel counts to “System -> Summary”
      • add comments to the generated sip, iax and extensions.conf files
      • add “Diagnostics -> Logs -> Calls” displays much more information now
      • add authentication is now optional in voicemail settings
      • update Asterisk to 1.4.5
      • bug fixed disk image packaging
      • bug fixed copy/paste bug in voicemail.inc
      • removed removed “cpu load” link from “System -> Summary”

      2007.06.14 – pb4

      • add “Dialplan -> Providers” page which lets one quickly map incoming extensions and phone permissions to providers
      • add “fromuser” and “fromdomain” fields to sip providers
      • add “top” links to status.php sections
      • add registrations to sip providers can now be disabled
      • add several operations have been greatly sped up with some additional configuration caching features
      • add zaptel device modules and utilities to root file system
      • add better checks and explanations for the network topology settings
      • refactored common display elements into functions
      • disk images now have a prefix of “pbx-” to avoid confusion with m0n0wall images
      • bug added a newline in sip.conf generator after port definition
      • bug stray asterisk bootup messages have been removed from “Interfaces -> Network” page
      • bug fixed some display issues when either no phones or no providers are present

      2007.06.08 – pb3

      • add some notes about codec selection
      • add some more start-up messages
      • bug starting asterisk with extra verbosity hangs, this is now done in two steps

      2007.06.07 – pb2

      • add iax provider and phone support
      • add “dtmfmode” option to sip providers and phones
      • add extensions.conf, sip.conf, iax.conf and voicemail.conf contents to status.php
      • add sip/iax2 show peers/registry output to status.php
      • add hints are now registered for sip phones
      • add lan gateway now configurable while setting lan ip on console
      • logging verbosity has been turned way up
      • bug deleted providers are now removed from phones which reference them
      • bug reverting to http in console now restarts mini httpd to update settings

      2007.06.01 – pb1

      • initial release