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637 lines
26 KiB
637 lines
26 KiB
NOTE: Corrections or additions to the ChangeLog may be submitted to
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http://bugs.digium.com. Documentation and formatting fixes are not
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not listed here. A complete listing of changes is available through
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the Asterisk-CVS mailing list hosted at http://lists.digium.com.
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Asterisk 1.0.10
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-- chan_local
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-- In releases 1.0.8 and 1.0.9, the Local channels that are created would
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not be masqueraded into the new channel type. This has now been fixed.
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-- chan_sip
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-- The 'insecure' options have been changed to support matching peersby IP
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only, not requiring authentication on incoming invites, or both. Before,
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to not require authentication on incoming invites also required matching
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peers based on IP only.
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-- chan_zap
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-- Before, call waiting could occur during the initial ringing on the line.
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This has now been fixed.
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-- app_disa
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-- We will now not set the accountcode if one is not supplied.
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-- app_meetme
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-- If the first caller into a conference hangs up while being prompted for
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the conference pin number, the conference will no longer be held open.
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-- app_userevent
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-- Events created with this application were indicated as a "call" event
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instead of a "user" event. This made the "user" event permissions
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not work correctly.
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-- app_voicemail
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-- When using the externpass option for voicemail, the password will be
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immediately updated in memory as well, instead of having to wait for
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the next time the configuration is reloaded.
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-- app_zapras
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-- We now ensure buffer policy is restored after RAS is done with a channel.
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This could cause audio problems on the channel after zapras is done
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with it.
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-- res_agi
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-- We now unmask the SIGHUP signal before executing an AGI script. This
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fixes problems where some AGI scripts would continue running long after
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the call is over.
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-- extensions
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-- A potential crash has been fixed when calling LEN() to get the length of
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a string that was 80 characters or larger.
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-- general
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-- Added man pages for astgenkey, autosupport, and safe_asterisk
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Asterisk 1.0.9
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-- fix bug in callerid matching in the dialplan that was introduced in 1.0.8
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Asterisk 1.0.8
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-- chan_zap
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-- Asterisk will now also look in the regular context for the fax extension
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while executing a macro. Previously, for this to work, the fax extension
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would have to be included in the macro definition.
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-- On some systems, ALERTING will be sent after PROCEEDING, so code has been
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added to account for this case.
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-- If no extension is specified on an overlap call, the 's' extension will
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be used.
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-- chan_sip
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-- We no longer send a "to" tag on "100 Trying" messages, as it is
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inappropriate to do so.
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-- We now respond correctly to an invite for T.38 with a "488 Not acceptable
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here"
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-- We now discard saved tags on 401/407 responses in case the provider we're
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talking to tries to pull a dirty trick on us and change it.
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-- rtptimeout options will now be correctly set on a peer basis rather than
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only global
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-- chan_mgcp
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-- Fixed setting of accountcode
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-- Fixed where *67 to block callerid only worked for first call
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-- chan_agent
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-- We now will not pass audio until the agent has acked the call if the
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configuration
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is set up for the agent to do so.
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-- chan_alsa
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-- Fixed problems with the unloading of this module
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-- res_agi
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-- A fix has been added to prevent calls from being hung up when more than
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one call is executing an AGI script calling the GET DATA command.
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-- AGI scripts will now continue to run even if a file was not found with
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the GET DATA command.
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-- When calling SAY NUMBER with a number like 09, we will now say "nine"
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instead of "zero"
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-- app_dial
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-- There was a problem where text frames would not be forwarded before the
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channel has been answered.
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-- app_disa
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-- Fixed the timeout used when no password is set
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-- app_queue
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-- Distinctive ring has been fixed to work for queue members
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-- rtp
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-- Fixed a logic error when setting the "rtpchecksums" option
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-- say.c
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-- A problem has been fixed with saying the date in Spanish.
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-- Makefile
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-- A line was missing for the autosupport script that caused "make rpm" to
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fail
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-- format_wav_gsm
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-- Fixed a problem with wav formatting that prevented files from being
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played in some media players
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-- pbx_spool
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-- Fixed if the last line of text in a file for the call spool did not
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contain a new line, it would not be processed
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-- logger
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-- Fixed the logger so that color escape sequences wouldn't be sent to the
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logs
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-- format_sln
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-- A lot of changes were made to correctly handle signed linear format on
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big endian machines
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-- asterisk.conf
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-- fix 'highpriority' option for asterisk.conf
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Asterisk 1.0.7
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-- chan_sip
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-- The fix for some codec availibility issues in 1.0.6 caused music on hold
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problems, but has now been fixed.
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-- chan_skinny
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-- A check has been added to avoid a crash.
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-- chan_iax2
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-- A feature has been added to CVS head to have the option of sending
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timestamps with trunk frames. It is not supported in 1.0, but a change
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has been made so that it will at least not choke if sent trunk
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timestamps.
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-- app_voicemail
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-- Some checks have been added to avoid a crash.
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-- speex
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-- The path /usr/include/speex has been added for a place to look for the
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speex header.
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Asterisk 1.0.6
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-- chan_iax2:
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-- Fixed a bug dealing with a division by zero that could cause a crash
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-- chan_sip:
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-- Behavior was changed so that when a registration fails due to DNS
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resolution issues, a retry will be attempted in 20 seconds.
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-- Peer settings were not reset to null values when reloading the
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configuration file. Behavior has been changed so that these values are
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now cleared.
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-- 'restrictcid' now properly works on MySQL peers.
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-- Only use the default callerid if it has been specified.
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-- Asterisk was not sending the same From: line in SIP messages during
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certain times. Fixed to make sure it stays the same. This makes some
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providers happier, to a working state.
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-- Certain circumstances involving a blank callerid caused asterisk to
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segmentation fault.
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-- There was a problem incorrectly matching codec availablity when global
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preferences were different from that of the user. To fix this,
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processing of SDP data has been moved to after determining who the call
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is coming from.
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-- Asterisk would run out of RTP ports while waiting for SUBSCRIBE's to
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expire even though an RTP port isn't needed in this case. This has been
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fixed by releasing the ports early.
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-- chan_zap:
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-- During a certain scenario when using flash and '#' transfers you would
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hear the other person and the music they were hearing. This has been
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fixed.
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-- A fix for a compilation issue with gcc4 was added.
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-- chan_modem_bestdata:
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-- A fix for a compilation issue with gcc4 was added.
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-- format_g729:
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-- Treat a 10-byte read as an end of file indication instead of an error.
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Some G729 encoders like to put 10-bytes at the end to indicate this.
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-- res_features:
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-- During certain situations when parking a call, both endpoints would get
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musiconhold. This has been fixed so the individual who parked the call
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will hear the digits and not musiconhold.
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-- app_dial:
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-- DIALEDPEERNUMBER is now being set, so if you attempted to use it in the
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past and failed, it should work now.
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-- A callerid change caused many headaches, this has been reversed to the
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original 1.0 behavior.
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-- A crash caused with the combination of the 'g' option and # transfer was
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fixed.
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-- app_voicemail:
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-- If two people hit the voicemail system at the same time, and were leaving
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a message the second message was overwriting the first. This has been
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fixed so that each one is distinct and will not overwrite eachother.
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-- cdr_tds:
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-- If the server you were using was going down, it had the potential to
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bring your asterisk server down with it. Extra stuff has been added so
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as to bring in more error/connection checking.
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-- cdr_pgsql:
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-- This will now attempt to reconnect after a connection problem.
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-- IAXY firmware:
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-- This has been updated to version 23. It includes a fix for lost
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registrations.
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-- internals
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-- Behavior was changed for 'show codec <number>' to make it more intuitive.
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-- DNS failures and asterisk do not get along too well, this is not totally
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the case anymore.
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-- Asterisk will now handle DNS failures at startup more gracefully, and
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won't crash and burn
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-- Choosing to append to a wave file would render the outputted wave file
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corrupt. Appending now works again.
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-- If you failed to define certain keys, asterisk had the potential to crash
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when seeing if you had used them.
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-- Attempting to use such things as ${EXTEN:-1} gave a wrong return value.
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However, this was never a documented feature...
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Asterisk 1.0.5
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-- chan_zap
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-- fix a callerid bug introduced in 1.0.4
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-- app_queue
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-- fix some penalty behavior
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Asterisk 1.0.4
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-- general
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-- fix memory leak evident with extensive use of variables
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-- update IAXy firmware to version 22
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-- enable some special write protection
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-- enable outbound DTMF
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-- fix seg fault with incorrect usage of SetVar
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-- other minor fixes including typos and doc updates
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-- chan_sip
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-- fix codecs to not be case sensitive
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-- Re-use auth credentials
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-- fix MWI when using type=friend
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-- fix global NAT option
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-- chan_agent / chan_local
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-- fix incorrect use count
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-- chan_zap
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-- Allow CID rings to be configured in zapata.conf
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-- no more patching needed for UK CID
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-- app_macro
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-- allow Macros to exit with '*' or '#' like regular extension processing
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-- app_voicemail
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-- don't allow '#' as a password
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-- add option to save voicemail before going to the operator
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-- fix global operator=yes
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-- app_read
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-- return 0 instead of -1 if user enters nothing
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-- res_agi
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-- don't exit AGI when file not found to stream
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-- send script parameter when using FastAGI
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Asterisk 1.0.3
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-- chan_zap
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-- fix seg fault when doing *0 to flash a trunk
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-- rtp
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-- seg fault fix
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-- chan_sip
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-- fix to prevent seg fault when attempting a transfer
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-- fix bug with supervised transfers
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-- fix codec preferences
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-- chan_h323
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-- fix compilation problem
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-- chan_iax2
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-- avoid a deadlock related to a static config of a BUNCH of peers
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-- cdr_pgsql
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-- fix memory leak when reading config
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-- Numerous other minor bug fixes
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Asterisk 1.0.2
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-- Major bugfix release
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Asterisk 1.0.1
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-- Added AGI over TCP support
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-- Add ability to purge callers from queue if no agents are logged in
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-- Fix inband PRI indication detection
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-- Fix for MGCP - always request digits if no RTP stream
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-- Fixed seg fault for ast_control_streamfile
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-- Make pick-up extension configurable via features.conf
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-- Numerous other bug fixes
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Asterisk 1.0.0
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-- Use Q.931 standard cause codes for asterisk cause codes
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-- Bug fixes from the bug tracker
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Asterisk 1.0-RC2
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-- Additional CDR backends
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-- Allow muted to reconnect
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-- Call parking improvements (including SIP parking support)
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-- Added licensed hold music from opsound.org
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-- GR-303 and Zap improvements
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-- More bug fixes from the bug tracker
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-- Improved FreeBSD/OpenBSD/MacOS X support
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Asterisk 1.0-RC1
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-- Innumerable bug fixes and features from the bug tracker
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-- Added Open Settlement Protocol (OSP) support
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-- Added Non-facility Associated Signalling (NFAS) Support
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-- Added alarm Monitoring support
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-- Added new MeetMe options
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-- Added GR-303 Support
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-- Added trunk groups
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-- ADPCM Standardization
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-- Numerous bug fixes
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-- Add IAX2 Firmware Support
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-- Add G.726 support
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-- Add ices/icecast support
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-- Numerous bug fixes
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Asterisk 0.7.2
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-- Countless small bug fixes from bug tracker
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-- DSP Fixes
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-- Fix unloading of Zaptel
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-- Pass Caller*ID/ANI properly on call forwarding
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-- Add indication for Italy
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Asterisk 0.7.1
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-- Fixed timed include context's and GotoIfTime
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-- Fixed chan_h323 it now gets remote ip properly instead of 127.0.0.1
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Asterisk 0.7.0
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-- Removed MP3 format and codec
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-- Can now load and unload SIP,IAX,IAX2,H323 channels without core
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-- Fixed various compiler warnings and clean up source tree
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-- Preliminary AES Support
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-- Fix SIP REINVITE
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-- Outbound SIP registration behind NAT using externip
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-- More CLI documentation and clean up
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-- Pin numbers on MeeMe
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-- Dynamic MeetMe conferences are more consistent with static conferences
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-- Added channel variables ${HANGUPCAUSE}, ${SIPDOMAIN}, ${TIMESTAMP}, ${ACCONTCODE}
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-- ODBC support for logging CDRs
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-- Indications for Norway and New Zeland
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-- Major redesign of app_voicemail
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-- Syslog support
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-- Reload logfiles with CLI command 'logger reload' and rotate logs with "logger rotate'
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-- Configurable DEBUG, NOTICE, WARNING, ERROR and ast_verbose messages now appear on remote console
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-- Properly reaping any zombie processes
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-- Added applications SayUnixTime, SetCDRUserField, HasNewVoicemail, ZapScan, Random, ResetCDR, NoCDR
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-- Make PRI Hangup Cause available to the dialplan
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-- Verify included contexts in extensions.conf
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-- Add DESTDIR support for building RPMs and packages
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-- Do route lookups on OpenBSD
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-- Add support for building on FreeBSD and OS X
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-- Add support for PostgreSQL in Voicemail
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-- Translate SIP hangup cause to PRI hangup cause where needed
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-- Better support for MOH in IAX2
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-- Fix SIP problem where channels were not removed on BYE
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-- Display codecs by name
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-- Remove MySQL and put PGSql instead for licensing reasons
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-- Better capability matching in SIP
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-- Full IBR4 compliance for chan_zap
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-- More flexible CDR handling
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-- Distinguish between BUSY and FAILURE on outbound calls
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-- Add initial support for SCCP via chan_skinny
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-- Better support for Future Group B signaling
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Asterisk 0.5.0
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-- Retain IAX2 and SIP registrations past shutdown/crash and restart
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-- True data mode bridging when possible
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-- H.323 build improvements
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-- Agent Callback-login support
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-- RFC2833 Improvements
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-- Add thread debugging
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-- Add optional pedantic SIP checking for Pingtel
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-- Allow extension names, include context, switch to use global vars.
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-- Allow variables in extensions.conf to reference previously defined ones
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-- Merge voicemail enhancements (app_voicemail2)
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-- Add multiple queueing strategies
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-- Merge support for 'T'
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-- Allow pending agent calling (Agent/:1)
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-- Add groupings to agents.conf
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-- Add video support to IAX2
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-- Zaptel optimize playback
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-- Add video support to SIP
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-- Make RTP ports configurable
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-- Add RDNIS support to SIP and IAX2
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-- Add transfer app (implement in SIP and IAX2)
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-- Make voicemail segmentable by context (app_voicemail2)
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-- Major restructuring of voicemail (app_voicemail2)
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-- Add initial ENUM support
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-- Add malloc debugging support
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-- Add preliminary Voicetronix support
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-- Add iLBC codec
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Asterisk 0.4.0
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-- Merge and edit Nick's FXO dial support
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-- Reengineer SIP registration (outbound)
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-- Support call pickup on SIP and compatibly with ZAP
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-- Support 302 Redirect on SIP
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-- Management interface improvements
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-- Add "hint" support
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-- Improve call forwarding using new "Local" channel driver.
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-- Add "Local" channel
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-- Substantial SIP enhancements including retransmissions
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-- Enforce case sensitivity on extension/context names
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-- Add monitor support (Thanks, Mahmut)
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-- Add experimental "trunk" option to IAX2 for high density VoIP
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-- Add experimental "debug channel" command
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-- Add 'C' flag to dial command to reset call detail record (handy for calling cards)
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-- Add NAT and dynamic support to MGCP
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-- Allow selection of in-band, out-of-band, or INFO based DTMF
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-- Add contributed "*80" support to blacklist numbers (Thanks James!)
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-- Add "NAT" option to sip user, peer, friend
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-- Add experimental "IAX2" protocol
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-- Change special variable "EXTEN-n" to "EXTEN:n" to follow Bash syntax
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-- Add "Enhanced" AGI with audio pass-through (voice recognition anyone?)
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-- Choose best priority from codec from allow/disallow
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-- Reject SIP calls to self
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-- Allow SIP registration to provide an alternative contact
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-- Make HOLD on SIP make use of asterisk MOH
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-- Add supervised transfer (tested with Pingtel only)
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-- Allow maxexpirey and defaultexpirey to be runtime configurable for SIP
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-- Preliminary codec 13 support (RFC3389)
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-- Add app_authenticate for general purpose authentication
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-- Optimize RTP and smoother
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-- Create special variable "EXTEN-n" where it is extension stripped by n MSD
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-- Fix uninitialized frame pointer in channel.c
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-- Add global variables support under [globals] of extensions.conf
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-- Add macro support (show application Macro)
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-- Allow [123-5] etc in extensions
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-- Allow format of App(arg1,arg2,...) instead of just App,arg1|arg2 in dialplan
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-- Add message waiting indicator to SIP
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-- Fix double free bug in channel.c
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Asterisk 0.3.0
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-- Add fastfoward, rewind, seek, and truncate functions to streams
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-- Support registration
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-- Add G729 format
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-- Permit applications to return a digit indicating new extension
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-- Change "SHUTDOWN" to "STOP" in commands
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-- SIP "Hold" fixes and VXML URI support
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-- New chan_zap with 160 sample chunk size
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-- Add DTMF, MF, and Fax tone detector to dsp routines
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-- Allow overlap dialing (inbound) on PRI
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-- Enable tone detection with PRI
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-- Add special information tone detection
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-- Add Asterisk DB support
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-- Add pulse dialing
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-- Re-record all system prompts
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-- Change "timelen" to samples for better accuracy
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-- Move to editline, eliminating readline dependency
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-- Add peer "poke" support to SIP and IAX
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-- Add experimental call progress detection
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-- Add SIP authentication (digest)
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-- Add RDNIS
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-- Reroute faxes to "fax" extension
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-- Create ISDN/modem group concept
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-- Centralize indication
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-- Add initial MGCP support
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-- SIP debugging cleanup
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-- SIP reload
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-- SIP commands (show channels, etc)
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-- Add optional busy detection
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-- Add Visual Message Waiting Indicator (MDMF and SDMF)
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-- Add ambiguous extension matching
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-- Add *69
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-- Major SIP enhancements from SIPit
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-- Rewrite of ZAP CLASS features using subchannels
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-- Enhanced call parking
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-- Add extended outgoing spool support (pbx_spool)
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Asterisk 0.2.0
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-- Outbound origination API
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-- Call management improvements
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-- Add Do Not Disturb (*78, *79)
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-- Add agents
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-- Document variables
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-- Add transfer capability on the console
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-- Add SpeeX codec translator
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-- Add call queues
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-- Add setcallerid functionality (AGI, application)
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-- Add special variables ${CALLERID}, ${EXTEN}, ${CONTEXT}, ${PRIORITY}
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-- Don't echo cancel on pure TDM connections by default
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-- Implement Async GOTO
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-- Differentiate softhangups
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-- Add date/time
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Asterisk 0.1.12
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-- Fix for Big Endian machines
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-- MySQL CDR Engine
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-- Various SIP fixes and enhancements
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-- Add "zapateller application and arbitrary tone pairs
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-- Don't always start at "s"
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-- Separate linear mode for pseudo and real
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-- Add initial RTP and SIP support (no jitter buffer yet, unknown stability)
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-- Add 'h' extension, executed on hangup
|
|
-- Add duration timer to message info
|
|
-- Add web based voicemail checking ("make webvmail")
|
|
-- Add ast_queue_frame function and eliminate frame pipes in most drivers
|
|
-- Centralize host access (and possibly future ACL's)
|
|
-- Add Caller*ID on PhoneJack (Thanks Nathan)
|
|
-- Add "safe_asterisk" wrapper script to auto-restart Asterisk
|
|
-- Indicate ringback on chan_phone
|
|
-- Add answer confirmation (press '#' to confirm answer)
|
|
-- Add distinctive ring support (e.g. Dial,Zap/4r2)
|
|
-- Add ANSI/vt100 color support
|
|
-- Make parking configurable through parking.conf
|
|
-- Fix the empty voicemail problem
|
|
-- Add Music On Hold
|
|
-- Add ADSI Compiler (app_adsiprog)
|
|
-- Extensive DISA re-work to improve tone generation
|
|
-- Reset all idle channels every 10 minutes on a PRI
|
|
-- Reset channels which are hungup with "channel in use"
|
|
-- Implement VNAK support in chan_iax
|
|
-- Fix chan_oss to support proper hangups and autoanswer
|
|
-- Make shutdown properly hangup channels
|
|
-- Add idling capability to chan_zap for idle-net
|
|
-- Add "MeetMe" conferencing app (app_meetme)
|
|
-- Add timing information to include
|
|
Asterisk 0.1.11
|
|
-- Add ISDN RAS capability
|
|
-- Add stutter dialtone to Chan Zap
|
|
-- Add "#include" capability to config files.
|
|
-- Add call-forward variable to Chan Zap (*72, *73)
|
|
-- Optimize IAX flow when transfer isn't possible
|
|
-- Allow transmission of ANI over IAX
|
|
Asterisk 0.1.10
|
|
-- Make ast_readstring parameter be the max # of digits, not the max size with \0
|
|
-- Make up any missing messages on the fly
|
|
-- Add support for specific DTMF interruption to saying numbers
|
|
-- Add new "u" and "b" options to condense busy/unavail handling
|
|
-- Add support for RSA authentication on IAX calls
|
|
-- Add support for ADSI compatible CPE
|
|
-- Outgoing call queue
|
|
-- Remote dialplan fixes for Quicknet
|
|
-- Added AGI commands supporting TDD functions (RECEIVE CHAR & TDD MODE)
|
|
-- Added TDD support (send/receive text in chan_zap)
|
|
-- Fix all strncpy references
|
|
-- Implement CSV CDR backend
|
|
-- Implement Call Detail Records
|
|
Asterisk 0.1.9
|
|
-- Implement IAX quelching
|
|
-- Allow Caller*ID to be overridden and suggested
|
|
-- Configure defaults to use IAXTEL
|
|
-- Allow remote dialplan polling via IAX
|
|
-- Eliminate ast_longest_extension
|
|
-- Implement dialplan request/reply
|
|
-- Let peers have allow/disallow for codecs
|
|
-- Change allow/deny to permit/deny in IAX
|
|
-- Allow dialplan entries to match Caller*ID as well
|
|
-- Added AGI (Asterisk Gateway Interface) scripting interface (app_agi)
|
|
-- Added chan_zap for zapata telephony kernel interface, removed chan_tor
|
|
-- Add convenience functions
|
|
-- Fix race condition in channel hangup
|
|
-- Fix memory leaks in both asterisk and iax frame allocations
|
|
-- Add "iax show stats" command and -DTRACE_FRAMES (for frame tracing)
|
|
-- Add DISA application (Thanks to Jim Dixon)
|
|
-- Add IAX transfer support
|
|
-- Add URL and HTML transmission
|
|
-- Add application for sending images
|
|
-- Add RedHat RPM spec file and build capability
|
|
-- Fix GSM WAV file format bug
|
|
-- Move ignorepat to main dialplan
|
|
-- Add ability to specificy TOS bits in IAX
|
|
-- Allow username:password in IAX strings
|
|
-- Updates to PhoneJack interface
|
|
-- Allow "servermail" in voicemail.conf to override e-mail in "from" line
|
|
-- Add 'skip' option to app_playback
|
|
-- Reject IAX calls on unknown extensions
|
|
-- Fix version stuff
|
|
Asterisk 0.1.8
|
|
-- Keep track of version information
|
|
-- Add -f to cause Asterisk not to fork
|
|
-- Keep important information in voicemail .txt file
|
|
-- Adtran Voice over Frame Relay updates
|
|
-- Implement option setting/querying of channel drivers
|
|
-- IAX performance improvements and protocol fixes
|
|
-- Substantial enhancement of console channel driver
|
|
-- Add IAX registration. Now IAX can dynamically register
|
|
-- Add flash-hook transfer on tormenta channels
|
|
-- Added Three Way Calling on tormenta channels
|
|
-- Start on concept of zombie channel
|
|
-- Add Call Waiting CallerID
|
|
-- Keep track of who registeres contexts, includes, and extensions
|
|
-- Added Call Waiting(tm), *67, *70, and *82 codes
|
|
-- Move parked calls into "parkedcalls" context by default
|
|
-- Allow dialplan to be displayed
|
|
-- Allow "=>" instead of just "=" to make instantiation clearer
|
|
-- Asterisk forks if called with no arguments
|
|
-- Add remote control by running asterisk -vvvc
|
|
-- Adjust verboseness with "set verbose" now
|
|
-- No longer requires libaudiofile
|
|
-- Install beep
|
|
-- Make PBX Config module reload extensions on SIGHUP
|
|
-- Allow modules to be reloaded when SIGHUP is received
|
|
-- Variables now contain line numbers
|
|
-- Make dialer send in band signalling
|
|
-- Add record application
|
|
-- Added PRI signalling to Tormenta driver
|
|
-- Allow use of BYEXTENSION in "Goto"
|
|
-- Allow adjustment of gains on tormenta channels
|
|
-- Added raw PCM file format support
|
|
-- Add U-law translator
|
|
-- Fix DTMF handling in bridge code
|
|
-- Fix access control with IAX
|
|
* Asterisk 0.1.7
|
|
-- Update configuration files and add some missing sounds
|
|
-- Added ability to include one context in another
|
|
-- Rewrite of PBX switching
|
|
-- Major mods to dialler application
|
|
-- Added Caller*ID spill reception
|
|
-- Added Dialogic VOX file format support
|
|
-- Added ADPCM Codec
|
|
-- Add Tormenta driver (RBS signalling)
|
|
-- Add Caller*ID spill creation
|
|
-- Rewrite of translation layer entirely
|
|
-- Add ability to run PBX without additional thread
|
|
* Asterisk 0.1.6
|
|
-- Make app_dial handle a lack of translators smoothly
|
|
-- Add ISDN4Linux support -- dtmf is weird...
|
|
-- Minor bug fixes
|
|
* Asterisk 0.1.5
|
|
-- Fix a small mistake in IAX
|
|
-- Fix the QuickNet driver to work with newer cards
|
|
* Asterisk 0.1.4
|
|
-- Update VoFR some more
|
|
-- Fix the QuickNet driver to work with LineJack
|
|
-- Add ability to pass images for IAX.
|
|
* Asterisk 0.1.3
|
|
-- Update VoFR for latest sangoma code
|
|
-- Update QuickNet Driver
|
|
-- Add text message handling
|
|
-- Fix transfers to use "default" if not in current context
|
|
-- Add call parking
|
|
-- Improve format/content negotiation
|
|
-- Added support for multiple languages
|
|
-- Bug fixes, as always...
|
|
* Asterisk 0.1.2
|
|
-- Updated README file with a "Getting Started" section
|
|
-- Added sample sounds and configuration files.
|
|
-- Added LPC10 very low bandwidth (low quality) compression
|
|
-- Enhanced translation selection mechanism.
|
|
-- Enhanced IAX jitter buffer, improved reliability
|
|
-- Support echo cancelation on PhoneJack
|
|
-- Updated PhoneJack driver to std. Telephony interface
|
|
-- Added app_echo for evaluating VoIP latency
|
|
-- Added app_system to execute arbitrary programs
|
|
-- Updated sample configuration files
|
|
-- Added OSS channel driver (full duplex only)
|
|
-- Added IAX implementation
|
|
-- Fixed some deadlocks.
|
|
-- A whole bunch of bug fixes
|
|
* Asterisk 0.1.1
|
|
-- Revised translator, fixed some general race conditions throughout *
|
|
-- Made dialer somewhat more aware of incompatible voice channels
|
|
-- Added Voice Modem driver and A/Open Modem Driver stub
|
|
-- Added MP3 decoder channel
|
|
-- Added Microsoft WAV49 support
|
|
-- Revised License -- Pure GPL, nothing else
|
|
-- Modified Copyright statement since code is still currently owned by author
|
|
-- Added RAW GSM headerless data format
|
|
-- Innumerable bug fixes
|
|
* Asterisk 0.1.0
|
|
-- Initial Release
|