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diff --git a/ChangeLog b/ChangeLog new file mode 100755 index 000000000..c8ca2cf67 --- /dev/null +++ b/ChangeLog @@ -0,0 +1,640 @@ + NOTE: Corrections or additions to the ChangeLog may be submitted to + http://bugs.digium.com. Documentation and formatting fixes are not + not listed here. A complete listing of changes is available through + the Asterisk-CVS mailing list hosted at http://lists.digium.com. + +Asterisk 1.0.10 + + -- chan_local + -- In releases 1.0.8 and 1.0.9, the Local channels that are created would + not be masqueraded into the new channel type. This has now been fixed. + -- chan_sip + -- The 'insecure' options have been changed to support matching peersby IP + only, not requiring authentication on incoming invites, or both. Before, + to not require authentication on incoming invites also required matching + peers based on IP only. + -- chan_zap + -- Before, call waiting could occur during the initial ringing on the line. + This has now been fixed. + -- app_disa + -- We will now not set the accountcode if one is not supplied. + -- app_meetme + -- If the first caller into a conference hangs up while being prompted for + the conference pin number, the conference will no longer be held open. + -- app_userevent + -- Events created with this application were indicated as a "call" event + instead of a "user" event. This made the "user" event permissions + not work correctly. + -- app_voicemail + -- When using the externpass option for voicemail, the password will be + immediately updated in memory as well, instead of having to wait for + the next time the configuration is reloaded. + -- app_zapras + -- We now ensure buffer policy is restored after RAS is done with a channel. + This could cause audio problems on the channel after zapras is done + with it. + -- res_agi + -- We now unmask the SIGHUP signal before executing an AGI script. This + fixes problems where some AGI scripts would continue running long after + the call is over. + -- extensions + -- A potential crash has been fixed when calling LEN() to get the length of + a string that was 80 characters or larger. + -- logger + -- The Asterisk logger will automatically detect when a log file needs to + be rotated. However, this feature could put Asterisk in a nasty loop + that would result in a crash. + -- general + -- Added man pages for astgenkey, autosupport, and safe_asterisk + +Asterisk 1.0.9 + + -- fix bug in callerid matching in the dialplan that was introduced in 1.0.8 + +Asterisk 1.0.8 + + -- chan_zap + -- Asterisk will now also look in the regular context for the fax extension + while executing a macro. Previously, for this to work, the fax extension + would have to be included in the macro definition. + -- On some systems, ALERTING will be sent after PROCEEDING, so code has been + added to account for this case. + -- If no extension is specified on an overlap call, the 's' extension will + be used. + -- chan_sip + -- We no longer send a "to" tag on "100 Trying" messages, as it is + inappropriate to do so. + -- We now respond correctly to an invite for T.38 with a "488 Not acceptable + here" + -- We now discard saved tags on 401/407 responses in case the provider we're + talking to tries to pull a dirty trick on us and change it. + -- rtptimeout options will now be correctly set on a peer basis rather than + only global + -- chan_mgcp + -- Fixed setting of accountcode + -- Fixed where *67 to block callerid only worked for first call + -- chan_agent + -- We now will not pass audio until the agent has acked the call if the + configuration + is set up for the agent to do so. + -- chan_alsa + -- Fixed problems with the unloading of this module + -- res_agi + -- A fix has been added to prevent calls from being hung up when more than + one call is executing an AGI script calling the GET DATA command. + -- AGI scripts will now continue to run even if a file was not found with + the GET DATA command. + -- When calling SAY NUMBER with a number like 09, we will now say "nine" + instead of "zero" + -- app_dial + -- There was a problem where text frames would not be forwarded before the + channel has been answered. + -- app_disa + -- Fixed the timeout used when no password is set + -- app_queue + -- Distinctive ring has been fixed to work for queue members + -- rtp + -- Fixed a logic error when setting the "rtpchecksums" option + -- say.c + -- A problem has been fixed with saying the date in Spanish. + -- Makefile + -- A line was missing for the autosupport script that caused "make rpm" to + fail + -- format_wav_gsm + -- Fixed a problem with wav formatting that prevented files from being + played in some media players + -- pbx_spool + -- Fixed if the last line of text in a file for the call spool did not + contain a new line, it would not be processed + -- logger + -- Fixed the logger so that color escape sequences wouldn't be sent to the + logs + -- format_sln + -- A lot of changes were made to correctly handle signed linear format on + big endian machines + -- asterisk.conf + -- fix 'highpriority' option for asterisk.conf + +Asterisk 1.0.7 + + -- chan_sip + -- The fix for some codec availibility issues in 1.0.6 caused music on hold + problems, but has now been fixed. + -- chan_skinny + -- A check has been added to avoid a crash. + -- chan_iax2 + -- A feature has been added to CVS head to have the option of sending + timestamps with trunk frames. It is not supported in 1.0, but a change + has been made so that it will at least not choke if sent trunk + timestamps. + -- app_voicemail + -- Some checks have been added to avoid a crash. + -- speex + -- The path /usr/include/speex has been added for a place to look for the + speex header. + +Asterisk 1.0.6 + + -- chan_iax2: + -- Fixed a bug dealing with a division by zero that could cause a crash + -- chan_sip: + -- Behavior was changed so that when a registration fails due to DNS + resolution issues, a retry will be attempted in 20 seconds. + -- Peer settings were not reset to null values when reloading the + configuration file. Behavior has been changed so that these values are + now cleared. + -- 'restrictcid' now properly works on MySQL peers. + -- Only use the default callerid if it has been specified. + -- Asterisk was not sending the same From: line in SIP messages during + certain times. Fixed to make sure it stays the same. This makes some + providers happier, to a working state. + -- Certain circumstances involving a blank callerid caused asterisk to + segmentation fault. + -- There was a problem incorrectly matching codec availablity when global + preferences were different from that of the user. To fix this, + processing of SDP data has been moved to after determining who the call + is coming from. + -- Asterisk would run out of RTP ports while waiting for SUBSCRIBE's to + expire even though an RTP port isn't needed in this case. This has been + fixed by releasing the ports early. + -- chan_zap: + -- During a certain scenario when using flash and '#' transfers you would + hear the other person and the music they were hearing. This has been + fixed. + -- A fix for a compilation issue with gcc4 was added. + -- chan_modem_bestdata: + -- A fix for a compilation issue with gcc4 was added. + -- format_g729: + -- Treat a 10-byte read as an end of file indication instead of an error. + Some G729 encoders like to put 10-bytes at the end to indicate this. + -- res_features: + -- During certain situations when parking a call, both endpoints would get + musiconhold. This has been fixed so the individual who parked the call + will hear the digits and not musiconhold. + -- app_dial: + -- DIALEDPEERNUMBER is now being set, so if you attempted to use it in the + past and failed, it should work now. + -- A callerid change caused many headaches, this has been reversed to the + original 1.0 behavior. + -- A crash caused with the combination of the 'g' option and # transfer was + fixed. + -- app_voicemail: + -- If two people hit the voicemail system at the same time, and were leaving + a message the second message was overwriting the first. This has been + fixed so that each one is distinct and will not overwrite eachother. + -- cdr_tds: + -- If the server you were using was going down, it had the potential to + bring your asterisk server down with it. Extra stuff has been added so + as to bring in more error/connection checking. + -- cdr_pgsql: + -- This will now attempt to reconnect after a connection problem. + -- IAXY firmware: + -- This has been updated to version 23. It includes a fix for lost + registrations. + -- internals + -- Behavior was changed for 'show codec <number>' to make it more intuitive. + -- DNS failures and asterisk do not get along too well, this is not totally + the case anymore. + -- Asterisk will now handle DNS failures at startup more gracefully, and + won't crash and burn + -- Choosing to append to a wave file would render the outputted wave file + corrupt. Appending now works again. + -- If you failed to define certain keys, asterisk had the potential to crash + when seeing if you had used them. + -- Attempting to use such things as ${EXTEN:-1} gave a wrong return value. + However, this was never a documented feature... + +Asterisk 1.0.5 + + -- chan_zap + -- fix a callerid bug introduced in 1.0.4 + -- app_queue + -- fix some penalty behavior + +Asterisk 1.0.4 + + -- general + -- fix memory leak evident with extensive use of variables + -- update IAXy firmware to version 22 + -- enable some special write protection + -- enable outbound DTMF + -- fix seg fault with incorrect usage of SetVar + -- other minor fixes including typos and doc updates + -- chan_sip + -- fix codecs to not be case sensitive + -- Re-use auth credentials + -- fix MWI when using type=friend + -- fix global NAT option + -- chan_agent / chan_local + -- fix incorrect use count + -- chan_zap + -- Allow CID rings to be configured in zapata.conf + -- no more patching needed for UK CID + -- app_macro + -- allow Macros to exit with '*' or '#' like regular extension processing + -- app_voicemail + -- don't allow '#' as a password + -- add option to save voicemail before going to the operator + -- fix global operator=yes + -- app_read + -- return 0 instead of -1 if user enters nothing + -- res_agi + -- don't exit AGI when file not found to stream + -- send script parameter when using FastAGI + +Asterisk 1.0.3 + + -- chan_zap + -- fix seg fault when doing *0 to flash a trunk + -- rtp + -- seg fault fix + -- chan_sip + -- fix to prevent seg fault when attempting a transfer + -- fix bug with supervised transfers + -- fix codec preferences + -- chan_h323 + -- fix compilation problem + -- chan_iax2 + -- avoid a deadlock related to a static config of a BUNCH of peers + -- cdr_pgsql + -- fix memory leak when reading config + -- Numerous other minor bug fixes + +Asterisk 1.0.2 + + -- Major bugfix release + +Asterisk 1.0.1 + + -- Added AGI over TCP support + -- Add ability to purge callers from queue if no agents are logged in + -- Fix inband PRI indication detection + -- 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 |