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authorrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2005-11-21 16:27:55 +0000
committerrussell <russell@f38db490-d61c-443f-a65b-d21fe96a405b>2005-11-21 16:27:55 +0000
commitdb9e060dd6d919b7257d7c8f05023cae19eaf6e1 (patch)
treef0cd5527d268946a83d2b2678f042e01c744070c
parent1e0f92c8b87f29ec1e11398df62c121bdaf49406 (diff)
remove the CHANGES file from v1-0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/v1-0@7175 f38db490-d61c-443f-a65b-d21fe96a405b
-rwxr-xr-xCHANGES638
1 files changed, 0 insertions, 638 deletions
diff --git a/CHANGES b/CHANGES
deleted file mode 100755
index de43197dd..000000000
--- a/CHANGES
+++ /dev/null
@@ -1,638 +0,0 @@
- 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.
-
- -- 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
- -- Fix for MGCP - always request digits if no RTP stream
- -- 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
-Asterisk 1.0-RC2
- -- Additional CDR backends
- -- Allow muted to reconnect
- -- Call parking improvements (including SIP parking support)
- -- Added licensed hold music from FreePlayMusic
- -- GR-303 and Zap improvements
- -- More bug fixes from the bug tracker
- -- Improved FreeBSD/OpenBSD/MacOS X support
-Asterisk 1.0-RC1
- -- Innumerable bug fixes and features from the bug tracker
- -- Added Open Settlement Protocol (OSP) support
- -- Added Non-facility Associated Signalling (NFAS) Support
- -- Added alarm Monitoring support
- -- Added new MeetMe options
- -- Added GR-303 Support
- -- Added trunk groups
- -- ADPCM Standardization
- -- Numerous bug fixes
- -- Add IAX2 Firmware Support
- -- Add G.726 support
- -- Add ices/icecast support
- -- Numerous bug fixes
-Asterisk 0.7.2
- -- Countless small bug fixes from bug tracker
- -- DSP Fixes
- -- Fix unloading of Zaptel
- -- Pass Caller*ID/ANI properly on call forwarding
- -- Add indication for Italy
-Asterisk 0.7.1
- -- Fixed timed include context's and GotoIfTime
- -- Fixed chan_h323 it now gets remote ip properly instead of 127.0.0.1
-Asterisk 0.7.0
- -- Removed MP3 format and codec
- -- Can now load and unload SIP,IAX,IAX2,H323 channels without core
- -- Fixed various compiler warnings and clean up source tree
- -- Preliminary AES Support
- -- Fix SIP REINVITE
- -- Outbound SIP registration behind NAT using externip
- -- More CLI documentation and clean up
- -- Pin numbers on MeeMe
- -- Dynamic MeetMe conferences are more consistent with static conferences
- -- Added channel variables ${HANGUPCAUSE}, ${SIPDOMAIN}, ${TIMESTAMP}, ${ACCONTCODE}
- -- ODBC support for logging CDRs
- -- Indications for Norway and New Zeland
- -- Major redesign of app_voicemail
- -- Syslog support
- -- Reload logfiles with CLI command 'logger reload' and rotate logs with "logger rotate'
- -- Configurable DEBUG, NOTICE, WARNING, ERROR and ast_verbose messages now appear on remote console
- -- Properly reaping any zombie processes
- -- Added applications SayUnixTime, SetCDRUserField, HasNewVoicemail, ZapScan, Random, ResetCDR, NoCDR
- -- Make PRI Hangup Cause available to the dialplan
- -- Verify included contexts in extensions.conf
- -- Add DESTDIR support for building RPMs and packages
- -- Do route lookups on OpenBSD
- -- Add support for building on FreeBSD and OS X
- -- Add support for PostgreSQL in Voicemail
- -- Translate SIP hangup cause to PRI hangup cause where needed
- -- Better support for MOH in IAX2
- -- Fix SIP problem where channels were not removed on BYE
- -- Display codecs by name
- -- Remove MySQL and put PGSql instead for licensing reasons
- -- Better capability matching in SIP
- -- Full IBR4 compliance for chan_zap
- -- More flexible CDR handling
- -- Distinguish between BUSY and FAILURE on outbound calls
- -- Add initial support for SCCP via chan_skinny
- -- Better support for Future Group B signaling
-Asterisk 0.5.0
- -- Retain IAX2 and SIP registrations past shutdown/crash and restart
- -- True data mode bridging when possible
- -- H.323 build improvements
- -- Agent Callback-login support
- -- RFC2833 Improvements
- -- Add thread debugging
- -- Add optional pedantic SIP checking for Pingtel
- -- Allow extension names, include context, switch to use global vars.
- -- Allow variables in extensions.conf to reference previously defined ones
- -- Merge voicemail enhancements (app_voicemail2)
- -- Add multiple queueing strategies
- -- Merge support for 'T'
- -- Allow pending agent calling (Agent/:1)
- -- Add groupings to agents.conf
- -- Add video support to IAX2
- -- Zaptel optimize playback
- -- Add video support to SIP
- -- Make RTP ports configurable
- -- Add RDNIS support to SIP and IAX2
- -- Add transfer app (implement in SIP and IAX2)
- -- Make voicemail segmentable by context (app_voicemail2)
- -- Major restructuring of voicemail (app_voicemail2)
- -- Add initial ENUM support
- -- Add malloc debugging support
- -- Add preliminary Voicetronix support
- -- Add iLBC codec
-Asterisk 0.4.0
- -- Merge and edit Nick's FXO dial support
- -- Reengineer SIP registration (outbound)
- -- Support call pickup on SIP and compatibly with ZAP
- -- Support 302 Redirect on SIP
- -- Management interface improvements
- -- Add "hint" support
- -- Improve call forwarding using new "Local" channel driver.
- -- Add "Local" channel
- -- Substantial SIP enhancements including retransmissions
- -- Enforce case sensitivity on extension/context names
- -- Add monitor support (Thanks, Mahmut)
- -- Add experimental "trunk" option to IAX2 for high density VoIP
- -- Add experimental "debug channel" command
- -- Add 'C' flag to dial command to reset call detail record (handy for calling cards)
- -- Add NAT and dynamic support to MGCP
- -- Allow selection of in-band, out-of-band, or INFO based DTMF
- -- Add contributed "*80" support to blacklist numbers (Thanks James!)
- -- Add "NAT" option to sip user, peer, friend
- -- Add experimental "IAX2" protocol
- -- Change special variable "EXTEN-n" to "EXTEN:n" to follow Bash syntax
- -- Add "Enhanced" AGI with audio pass-through (voice recognition anyone?)
- -- Choose best priority from codec from allow/disallow
- -- Reject SIP calls to self
- -- Allow SIP registration to provide an alternative contact
- -- Make HOLD on SIP make use of asterisk MOH
- -- Add supervised transfer (tested with Pingtel only)
- -- Allow maxexpirey and defaultexpirey to be runtime configurable for SIP
- -- Preliminary codec 13 support (RFC3389)
- -- Add app_authenticate for general purpose authentication
- -- Optimize RTP and smoother
- -- Create special variable "EXTEN-n" where it is extension stripped by n MSD
- -- Fix uninitialized frame pointer in channel.c
- -- Add global variables support under [globals] of extensions.conf
- -- Add macro support (show application Macro)
- -- Allow [123-5] etc in extensions
- -- Allow format of App(arg1,arg2,...) instead of just App,arg1|arg2 in dialplan
- -- Add message waiting indicator to SIP
- -- Fix double free bug in channel.c
-Asterisk 0.3.0
- -- Add fastfoward, rewind, seek, and truncate functions to streams
- -- Support registration
- -- Add G729 format
- -- Permit applications to return a digit indicating new extension
- -- Change "SHUTDOWN" to "STOP" in commands
- -- SIP "Hold" fixes and VXML URI support
- -- New chan_zap with 160 sample chunk size
- -- Add DTMF, MF, and Fax tone detector to dsp routines
- -- Allow overlap dialing (inbound) on PRI
- -- Enable tone detection with PRI
- -- Add special information tone detection
- -- Add Asterisk DB support
- -- Add pulse dialing
- -- Re-record all system prompts
- -- Change "timelen" to samples for better accuracy
- -- Move to editline, eliminating readline dependency
- -- Add peer "poke" support to SIP and IAX
- -- Add experimental call progress detection
- -- Add SIP authentication (digest)
- -- Add RDNIS
- -- Reroute faxes to "fax" extension
- -- Create ISDN/modem group concept
- -- Centralize indication
- -- Add initial MGCP support
- -- SIP debugging cleanup
- -- SIP reload
- -- SIP commands (show channels, etc)
- -- Add optional busy detection
- -- Add Visual Message Waiting Indicator (MDMF and SDMF)
- -- Add ambiguous extension matching
- -- Add *69
- -- Major SIP enhancements from SIPit
- -- Rewrite of ZAP CLASS features using subchannels
- -- Enhanced call parking
- -- Add extended outgoing spool support (pbx_spool)
-Asterisk 0.2.0
- -- Outbound origination API
- -- Call management improvements
- -- Add Do Not Disturb (*78, *79)
- -- Add agents
- -- Document variables
- -- Add transfer capability on the console
- -- Add SpeeX codec translator
- -- Add call queues
- -- Add setcallerid functionality (AGI, application)
- -- Add special variables ${CALLERID}, ${EXTEN}, ${CONTEXT}, ${PRIORITY}
- -- Don't echo cancel on pure TDM connections by default
- -- Implement Async GOTO
- -- Differentiate softhangups
- -- Add date/time
-Asterisk 0.1.12
- -- Fix for Big Endian machines
- -- MySQL CDR Engine
- -- Various SIP fixes and enhancements
- -- Add "zapateller application and arbitrary tone pairs
- -- Don't always start at "s"
- -- Separate linear mode for pseudo and real
- -- Add initial RTP and SIP support (no jitter buffer yet, unknown stability)
- -- 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