diff options
| -rwxr-xr-x | CHANGES | 638 |
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 |