@ -1,21 +1,286 @@
-- Pass redirecting number on PRI calls
-- Add RTP debug support
-- Misc Debugging improvements
-- Add TALK_DETECTED variable
-- Adding Q.SIG switchtype option to chan_zap
-- Added pbx_builtin_serialize_variables
-- Update to new iLBC codec
-- Add CLI for realtime stuff
-- Add DUNDi.... (http://www.dundi.com)
-- Misc Memory fixes
-- Voicemail improvements from the bug tracker
-- Major revamp of PBX core including 'n' and 's' priorities and labels
-- Deprecate pbx_wilcalu and app_qcall in favor of pbx_spool
-- Remove old chan_iax and chan_vofr
-- Major Caller*ID Restructuring
-- Realtime API (IAX, SIP and Voicemail)
-- codecs.conf to tune various codec options (ie Speex)
NOTE: Corrections or additions to the ChangeLog may be submitted to
http://bugs.digium.com. Documentation and formatting fixes are not
not listed here. A complete listing of changes is available through
the Asterisk-CVS mailing list hosted at http://lists.digium.com.
Asterisk 1.2.0
-- Some of the major feature upgrades ...
-- DUNDi (Distributed Universal Number Discovery -- http://www.dundi.com)
-- AEL (Asterisk Extension Logic)
-- Realtime Database Configuration Engine
-- Native Music on Hold
-- Native IAX Encryption
-- New Jitter Buffer
-- Q.SIG Switchtype for PRI
-- FastAGI (AGI over TCP)
-- Dialplan Functions
-- ODBC Storage of Voicemail
Asterisk 1.0.10
-- chan_local
-- In releases 1.0.8 and 1.0.9, the Local channels that are created would
not be masqueraded into the new channel type. This has now been fixed.
-- chan_sip
-- The 'insecure' options have been changed to support matching peersby IP
only, not requiring authentication on incoming invites, or both. Before,
to not require authentication on incoming invites also required matching
peers based on IP only.
-- chan_zap
-- Before, call waiting could occur during the initial ringing on the line.
This has now been fixed.
-- app_disa
-- We will now not set the accountcode if one is not supplied.
-- app_meetme
-- If the first caller into a conference hangs up while being prompted for
the conference pin number, the conference will no longer be held open.
-- app_userevent
-- Events created with this application were indicated as a "call" event
instead of a "user" event. This made the "user" event permissions
not work correctly.
-- app_voicemail
-- When using the externpass option for voicemail, the password will be
immediately updated in memory as well, instead of having to wait for
the next time the configuration is reloaded.
-- app_zapras
-- We now ensure buffer policy is restored after RAS is done with a channel.
This could cause audio problems on the channel after zapras is done
with it.
-- res_agi
-- We now unmask the SIGHUP signal before executing an AGI script. This
fixes problems where some AGI scripts would continue running long after
the call is over.
-- extensions
-- A potential crash has been fixed when calling LEN() to get the length of
a string that was 80 characters or larger.
-- logger
-- The Asterisk logger will automatically detect when a log file needs to
be rotated. However, this feature could put Asterisk in a nasty loop
that would result in a crash.
-- general
-- Added man pages for astgenkey, autosupport, and safe_asterisk
Asterisk 1.0.9
-- fix bug in callerid matching in the dialplan that was introduced in 1.0.8
Asterisk 1.0.8
-- chan_zap
-- Asterisk will now also look in the regular context for the fax extension
while executing a macro. Previously, for this to work, the fax extension
would have to be included in the macro definition.
-- On some systems, ALERTING will be sent after PROCEEDING, so code has been
added to account for this case.
-- If no extension is specified on an overlap call, the 's' extension will
be used.
-- chan_sip
-- We no longer send a "to" tag on "100 Trying" messages, as it is
inappropriate to do so.
-- We now respond correctly to an invite for T.38 with a "488 Not acceptable
here"
-- We now discard saved tags on 401/407 responses in case the provider we're
talking to tries to pull a dirty trick on us and change it.
-- rtptimeout options will now be correctly set on a peer basis rather than
only global
-- chan_mgcp
-- Fixed setting of accountcode
-- Fixed where *67 to block callerid only worked for first call
-- chan_agent
-- We now will not pass audio until the agent has acked the call if the
configuration
is set up for the agent to do so.
-- chan_alsa
-- Fixed problems with the unloading of this module
-- res_agi
-- A fix has been added to prevent calls from being hung up when more than
one call is executing an AGI script calling the GET DATA command.
-- AGI scripts will now continue to run even if a file was not found with
the GET DATA command.
-- When calling SAY NUMBER with a number like 09, we will now say "nine"
instead of "zero"
-- app_dial
-- There was a problem where text frames would not be forwarded before the
channel has been answered.
-- app_disa
-- Fixed the timeout used when no password is set
-- app_queue
-- Distinctive ring has been fixed to work for queue members
-- rtp
-- Fixed a logic error when setting the "rtpchecksums" option
-- say.c
-- A problem has been fixed with saying the date in Spanish.
-- Makefile
-- A line was missing for the autosupport script that caused "make rpm" to
fail
-- format_wav_gsm
-- Fixed a problem with wav formatting that prevented files from being
played in some media players
-- pbx_spool
-- Fixed if the last line of text in a file for the call spool did not
contain a new line, it would not be processed
-- logger
-- Fixed the logger so that color escape sequences wouldn't be sent to the
logs
-- format_sln
-- A lot of changes were made to correctly handle signed linear format on
big endian machines
-- asterisk.conf
-- fix 'highpriority' option for asterisk.conf
Asterisk 1.0.7
-- chan_sip
-- The fix for some codec availibility issues in 1.0.6 caused music on hold
problems, but has now been fixed.
-- chan_skinny
-- A check has been added to avoid a crash.
-- chan_iax2
-- A feature has been added to CVS head to have the option of sending
timestamps with trunk frames. It is not supported in 1.0, but a change
has been made so that it will at least not choke if sent trunk
timestamps.
-- app_voicemail
-- Some checks have been added to avoid a crash.
-- speex
-- The path /usr/include/speex has been added for a place to look for the
speex header.
Asterisk 1.0.6
-- chan_iax2:
-- Fixed a bug dealing with a division by zero that could cause a crash
-- chan_sip:
-- Behavior was changed so that when a registration fails due to DNS
resolution issues, a retry will be attempted in 20 seconds.
-- Peer settings were not reset to null values when reloading the
configuration file. Behavior has been changed so that these values are
now cleared.
-- 'restrictcid' now properly works on MySQL peers.
-- Only use the default callerid if it has been specified.
-- Asterisk was not sending the same From: line in SIP messages during
certain times. Fixed to make sure it stays the same. This makes some
providers happier, to a working state.
-- Certain circumstances involving a blank callerid caused asterisk to
segmentation fault.
-- There was a problem incorrectly matching codec availablity when global
preferences were different from that of the user. To fix this,
processing of SDP data has been moved to after determining who the call
is coming from.
-- Asterisk would run out of RTP ports while waiting for SUBSCRIBE's to
expire even though an RTP port isn't needed in this case. This has been
fixed by releasing the ports early.
-- chan_zap:
-- During a certain scenario when using flash and '#' transfers you would
hear the other person and the music they were hearing. This has been
fixed.
-- A fix for a compilation issue with gcc4 was added.
-- chan_modem_bestdata:
-- A fix for a compilation issue with gcc4 was added.
-- format_g729:
-- Treat a 10-byte read as an end of file indication instead of an error.
Some G729 encoders like to put 10-bytes at the end to indicate this.
-- res_features:
-- During certain situations when parking a call, both endpoints would get
musiconhold. This has been fixed so the individual who parked the call
will hear the digits and not musiconhold.
-- app_dial:
-- DIALEDPEERNUMBER is now being set, so if you attempted to use it in the
past and failed, it should work now.
-- A callerid change caused many headaches, this has been reversed to the
original 1.0 behavior.
-- A crash caused with the combination of the 'g' option and # transfer was
fixed.
-- app_voicemail:
-- If two people hit the voicemail system at the same time, and were leaving
a message the second message was overwriting the first. This has been
fixed so that each one is distinct and will not overwrite eachother.
-- cdr_tds:
-- If the server you were using was going down, it had the potential to
bring your asterisk server down with it. Extra stuff has been added so
as to bring in more error/connection checking.
-- cdr_pgsql:
-- This will now attempt to reconnect after a connection problem.
-- IAXY firmware:
-- This has been updated to version 23. It includes a fix for lost
registrations.
-- internals
-- Behavior was changed for 'show codec <number>' to make it more intuitive.
-- DNS failures and asterisk do not get along too well, this is not totally
the case anymore.
-- Asterisk will now handle DNS failures at startup more gracefully, and
won't crash and burn
-- Choosing to append to a wave file would render the outputted wave file
corrupt. Appending now works again.
-- If you failed to define certain keys, asterisk had the potential to crash
when seeing if you had used them.
-- Attempting to use such things as ${EXTEN:-1} gave a wrong return value.
However, this was never a documented feature...
Asterisk 1.0.5
-- chan_zap
-- fix a callerid bug introduced in 1.0.4
-- app_queue
-- fix some penalty behavior
Asterisk 1.0.4
-- general
-- fix memory leak evident with extensive use of variables
-- update IAXy firmware to version 22
-- enable some special write protection
-- enable outbound DTMF
-- fix seg fault with incorrect usage of SetVar
-- other minor fixes including typos and doc updates
-- chan_sip
-- fix codecs to not be case sensitive
-- Re-use auth credentials
-- fix MWI when using type=friend
-- fix global NAT option
-- chan_agent / chan_local
-- fix incorrect use count
-- chan_zap
-- Allow CID rings to be configured in zapata.conf
-- no more patching needed for UK CID
-- app_macro
-- allow Macros to exit with '*' or '#' like regular extension processing
-- app_voicemail
-- don't allow '#' as a password
-- add option to save voicemail before going to the operator
-- fix global operator=yes
-- app_read
-- return 0 instead of -1 if user enters nothing
-- res_agi
-- don't exit AGI when file not found to stream
-- send script parameter when using FastAGI
Asterisk 1.0.3
-- chan_zap
-- fix seg fault when doing *0 to flash a trunk
-- rtp
-- seg fault fix
-- chan_sip
-- fix to prevent seg fault when attempting a transfer
-- fix bug with supervised transfers
-- fix codec preferences
-- chan_h323
-- fix compilation problem
-- chan_iax2
-- avoid a deadlock related to a static config of a BUNCH of peers
-- cdr_pgsql
-- fix memory leak when reading config
-- Numerous other minor bug fixes
Asterisk 1.0.2
-- Major bugfix release
Asterisk 1.0.1
-- Added AGI over TCP support
-- Add ability to purge callers from queue if no agents are logged in
-- Fix inband PRI indication detection
@ -23,6 +288,7 @@ Asterisk 1.0.1
-- Fixed seg fault for ast_control_streamfile
-- Make pick-up extension configurable via features.conf
-- Numerous other bug fixes
Asterisk 1.0.0
-- Use Q.931 standard cause codes for asterisk cause codes
-- Bug fixes from the bug tracker
Russell Bryant
20 years ago