Askozia PBX phone system

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

Changelog

Changelog – 2.2 Series

2013.05.07 – 2.2.3

2012.12.14 – 2.2.2

2012.12.14 – 2.2.2CFE

2012.11.09 – 2.2.1

2012.11.09 – 2.2.1CFE

2012.08.15 – 2.2CFE

2012.08.15 – 2.2

2012.08.15 – 2.1.7CFE

2012.08.15 – 2.1.7

2012.04.26 – 2.1.6CFE

2012.03.09 – 2.1.6

2011.12.23 – 2.1.5

2011.12.02 – 2.1.4

2011.07.29 – 2.1.3

internal release – 2.1.2

2011.07.07 – 2.1.1

2011.06.24 – 2.1

2011.04.11 – 2.0.4

2010.11.26 – 2.0.3

2010.10.22 – 2.0.2

2010.06.03 – 2.0.1

2010.05.04 – 2.0.0

2010.04.09 – 2.0.rc3

2010.03.12 – 2.0.rc2

2010.02.13 – 2.0.rc1

1.0 Series

2009.05.29 – 1.0.3

2009.03.10 – 1.0.2

2008.09.19 – 1.0.1

2008.09.12 – 1.0.0

2008.08.21 – pb14.3

2008.08.15 – pb14.2

2008.08.08 – pb14.1

2008.07.31 – pb14

2008.06.05 – pb13.4

2008.03.27 – pb13.3

2008.03.12 – pb13.2

2008.02.26 – pb13.1

2008.02.14 – pb13

2007.11.16 – pb12.2

2007.11.09 – pb12.1

2007.11.09 – pb12

2007.10.09 – pb11.1

2007.10.05 – pb11

2007.09.07 – pb10

2007.08.09 – pb9

2007.08.02 – pb8

2007.07.26 – pb7

2007.07.05 – pb6

2007.06.21 – pb5

2007.06.14 – pb4

2007.06.08 – pb3

2007.06.07 – pb2

2007.06.01 – pb1

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