2009.05.29 – 1.0.3

• French localization
• Japanese voicemail template
• updated German localization

2009.03.10 – 1.0.2

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

2008.09.19 – 1.0.1

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

2008.09.12 – 1.0.0

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

2008.08.21 – pb14.3

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

2008.08.15 – pb14.2

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

2008.08.08 – pb14.1

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

2008.07.31 – pb14

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

2008.06.05 – pb13.4

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

2008.03.27 – pb13.3

• isdn and analog interface settings can now be “forgotten” in the webGUI
• dialplan now produces more human-readable log messages
• Page() application to base distribution
• manual attributes can now be defined for ISDN interfaces
• “readback” numbers (used for unreachable messages) can now be set manually for SIP and IAX providers
• Voicemail enabled extensions now have an option to signal “busy” via tones instead of going to Voicemail
• 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
• incoming calls from ISDN providers will now be accepted in more cases
• ISDN and Analog interfaces are now automatically renamed from “(unconfigured)” upon configuration
• ISDN Operating Mode is now verified before saving
• missed call notifications are no longer sent when a voicemail message was left
• using applications as a Provider’s incoming destination works properly again (potentially still not working) (reported by Marco in the forums)
• outgoing SIP uri dialing logic has been simplified
• main macro completely rewritten

2008.03.12 – pb13.2

• transmit and receive gains can now be set for analog interfaces (working patch provided by devon in the forums, modified for code consistency)
• improved documentation on the analog and isdn interfaces pages
• manual attributes can now be defined for analog interfaces
• an authentication method can now be selected for the SMTP server used in “Services -> Voicemail”
• * and # characters may now be used in application extensions
• missed call notifications are no longer sent for successfully completed calls
• 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
• 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)
• echo squelch options removed from isdn interfaces as it is no longer supported in isdn4bsd

2008.02.26 – pb13.1

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

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

2007.11.16 – pb12.2

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

2007.11.09 – pb12.1

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

2007.11.09 – pb12

• PC Engines – ALIX.2,3x
• outgoing caller id override options to SIP and IAX providers
• English (UK) ulaw prompts, renamed existing prompts to English (USA)
• French (France) gsm prompts, renamed existing prompts to French (Canada)
• missed call notification e-mail option to phones
• chan_local jitterbuffer patch, enabling applications to also be jitterbuffered
• hints to call parking spots based on info provided by Mat M
• multiple incoming extensions per provider can now be defined
• multiple, individually addressable, ISDN phones may now be connected to a single port
• 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
• fixed “localnet” setting in sip.conf
• fixed snooze feature in WakeMe
• “Diagnostics -> Logs -> Calls” now displays information in the “src” field in more cases
• fixed a small display issue when a provider does not have any patterns defined
• unconfigured analog interfaces now have a default value of “128″ for their echo cancellers instead of “yes”
• removed gsm prompts from languages which also have ulaw prompts to make room for more languages
• 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

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

2007.10.05 – pb11

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

2007.09.07 – pb10

• Polish voicemail notification e-mail translation
• Russian language (gsm) audio prompts
• MAC spoofing support for the network interface
• “attended transfer answer,” “transfer key” and “extension digit” timeout options to “Dialplan -> Transfers”
• collapsable “Advanced” menu and option to keep it open in “Advanced -> GUI Options”
• options in “Advanced -> GUI Options” to hide menu entries for unused telephony technologies
• call-limit and busy-limit options to SIP Phones and fixed SIP hints in dialplan, basic presence information is now working
• WakeMe – Wake-Up Call Manager to extension 00009253 (0000WAKE)
• a basic printable Dialplan
• “Advanced -> RTP” page so the RTP port range can be defined
• initial Analog Phone support
• debugging tools for USB ISDN devices
• 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
• Asterisk 1.4.11
• 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
• fixed the broken jitterbuffer options on in “Advanced -> IAX”
• fixed sound issues introduced in pb9 in certain situations (IAX channels are still having some problems)
• incoming extension reference deletion is now centralized and used in previously omitted cases
• configured, but absent analog interfaces are no longer initialized (thus failing) during boot and during Asterisk restarts
• ab-units are no longer displayed on “Interfaces -> Network -> Assign”

2007.08.09 – pb9

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

2007.08.02 – pb8

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

2007.07.26 – pb7

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

2007.07.05 – pb6

• improved sound packaging to finally include only what is necessary
• 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
• help messages now only appear once per page when multiple instances of the same field are present
• phones and providers appear properly sorted now (earlier first appeared sorted SIP then sorted IAX)
• multiple patterns may now be entered
• audio prompts for the following languages:
  • Dutch (gsm)
  • French (gsm/ulaw)
  • German (gsm)
  • Italian (gsm)
  • Japanese (gsm)
  • Spanish (gsm/ulaw)
  • Swedish (gsm)
• registration timeout options to sip providers
• parallel ringing call group support
• iLBC and Speex codecs
• FreeBSD 6.2-RELEASE-p5
• Asterisk to 1.4.6
• normalized some pages’ POST and redirection routines
• removed prefixes (automatically converted to patterns on the first boot)

2007.06.21 – pb5

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

2007.06.14 – pb4

• “Dialplan -> Providers” page which lets one quickly map incoming extensions and phone permissions to providers
• “fromuser” and “fromdomain” fields to sip providers
• “top” links to status.php sections
• registrations to sip providers can now be disabled
• several operations have been greatly sped up with some additional configuration caching features
• zaptel device modules and utilities to root file system
• 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
• added a newline in sip.conf generator after port definition
• stray asterisk bootup messages have been removed from “Interfaces -> Network” page
• fixed some display issues when either no phones or no providers are present

2007.06.08 – pb3

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

2007.06.07 – pb2

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

2007.06.01 – pb1

• initial release