diff options
Diffstat (limited to 'CHANGES')
| -rwxr-xr-x | CHANGES | 300 |
1 files changed, 283 insertions, 17 deletions
@@ -1,21 +1,286 @@ - -- Pass redirecting number on PRI calls - -- Add RTP debug support - -- Misc Debugging improvements - -- Add TALK_DETECTED variable - -- Adding Q.SIG switchtype option to chan_zap - -- Added pbx_builtin_serialize_variables - -- Update to new iLBC codec - -- Add CLI for realtime stuff - -- Add DUNDi.... (http://www.dundi.com) - -- Misc Memory fixes - -- Voicemail improvements from the bug tracker - -- Major revamp of PBX core including 'n' and 's' priorities and labels - -- Deprecate pbx_wilcalu and app_qcall in favor of pbx_spool - -- Remove old chan_iax and chan_vofr - -- Major Caller*ID Restructuring - -- Realtime API (IAX, SIP and Voicemail) - -- codecs.conf to tune various codec options (ie Speex) + NOTE: Corrections or additions to the ChangeLog may be submitted to + http://bugs.digium.com. Documentation and formatting fixes are not + not listed here. A complete listing of changes is available through + the Asterisk-CVS mailing list hosted at http://lists.digium.com. + +Asterisk 1.2.0 + + -- Some of the major feature upgrades ... + + -- DUNDi (Distributed Universal Number Discovery -- http://www.dundi.com) + -- AEL (Asterisk Extension Logic) + -- Realtime Database Configuration Engine + -- Native Music on Hold + -- Native IAX Encryption + -- New Jitter Buffer + -- Q.SIG Switchtype for PRI + -- FastAGI (AGI over TCP) + -- Dialplan Functions + -- ODBC Storage of Voicemail + +Asterisk 1.0.10 + + -- chan_local + -- In releases 1.0.8 and 1.0.9, the Local channels that are created would + not be masqueraded into the new channel type. This has now been fixed. + -- chan_sip + -- The 'insecure' options have been changed to support matching peersby IP + only, not requiring authentication on incoming invites, or both. Before, + to not require authentication on incoming invites also required matching + peers based on IP only. + -- chan_zap + -- Before, call waiting could occur during the initial ringing on the line. + This has now been fixed. + -- app_disa + -- We will now not set the accountcode if one is not supplied. + -- app_meetme + -- If the first caller into a conference hangs up while being prompted for + the conference pin number, the conference will no longer be held open. + -- app_userevent + -- Events created with this application were indicated as a "call" event + instead of a "user" event. This made the "user" event permissions + not work correctly. + -- app_voicemail + -- When using the externpass option for voicemail, the password will be + immediately updated in memory as well, instead of having to wait for + the next time the configuration is reloaded. + -- app_zapras + -- We now ensure buffer policy is restored after RAS is done with a channel. + This could cause audio problems on the channel after zapras is done + with it. + -- res_agi + -- We now unmask the SIGHUP signal before executing an AGI script. This + fixes problems where some AGI scripts would continue running long after + the call is over. + -- extensions + -- A potential crash has been fixed when calling LEN() to get the length of + a string that was 80 characters or larger. + -- logger + -- The Asterisk logger will automatically detect when a log file needs to + be rotated. However, this feature could put Asterisk in a nasty loop + that would result in a crash. + -- general + -- Added man pages for astgenkey, autosupport, and safe_asterisk + +Asterisk 1.0.9 + + -- fix bug in callerid matching in the dialplan that was introduced in 1.0.8 + +Asterisk 1.0.8 + + -- chan_zap + -- Asterisk will now also look in the regular context for the fax extension + while executing a macro. Previously, for this to work, the fax extension + would have to be included in the macro definition. + -- On some systems, ALERTING will be sent after PROCEEDING, so code has been + added to account for this case. + -- If no extension is specified on an overlap call, the 's' extension will + be used. + -- chan_sip + -- We no longer send a "to" tag on "100 Trying" messages, as it is + inappropriate to do so. + -- We now respond correctly to an invite for T.38 with a "488 Not acceptable + here" + -- We now discard saved tags on 401/407 responses in case the provider we're + talking to tries to pull a dirty trick on us and change it. + -- rtptimeout options will now be correctly set on a peer basis rather than + only global + -- chan_mgcp + -- Fixed setting of accountcode + -- Fixed where *67 to block callerid only worked for first call + -- chan_agent + -- We now will not pass audio until the agent has acked the call if the + configuration + is set up for the agent to do so. + -- chan_alsa + -- Fixed problems with the unloading of this module + -- res_agi + -- A fix has been added to prevent calls from being hung up when more than + one call is executing an AGI script calling the GET DATA command. + -- AGI scripts will now continue to run even if a file was not found with + the GET DATA command. + -- When calling SAY NUMBER with a number like 09, we will now say "nine" + instead of "zero" + -- app_dial + -- There was a problem where text frames would not be forwarded before the + channel has been answered. + -- app_disa + -- Fixed the timeout used when no password is set + -- app_queue + -- Distinctive ring has been fixed to work for queue members + -- rtp + -- Fixed a logic error when setting the "rtpchecksums" option + -- say.c + -- A problem has been fixed with saying the date in Spanish. + -- Makefile + -- A line was missing for the autosupport script that caused "make rpm" to + fail + -- format_wav_gsm + -- Fixed a problem with wav formatting that prevented files from being + played in some media players + -- pbx_spool + -- Fixed if the last line of text in a file for the call spool did not + contain a new line, it would not be processed + -- logger + -- Fixed the logger so that color escape sequences wouldn't be sent to the + logs + -- format_sln + -- A lot of changes were made to correctly handle signed linear format on + big endian machines + -- asterisk.conf + -- fix 'highpriority' option for asterisk.conf + +Asterisk 1.0.7 + + -- chan_sip + -- The fix for some codec availibility issues in 1.0.6 caused music on hold + problems, but has now been fixed. + -- chan_skinny + -- A check has been added to avoid a crash. + -- chan_iax2 + -- A feature has been added to CVS head to have the option of sending + timestamps with trunk frames. It is not supported in 1.0, but a change + has been made so that it will at least not choke if sent trunk + timestamps. + -- app_voicemail + -- Some checks have been added to avoid a crash. + -- speex + -- The path /usr/include/speex has been added for a place to look for the + speex header. + +Asterisk 1.0.6 + + -- chan_iax2: + -- Fixed a bug dealing with a division by zero that could cause a crash + -- chan_sip: + -- Behavior was changed so that when a registration fails due to DNS + resolution issues, a retry will be attempted in 20 seconds. + -- Peer settings were not reset to null values when reloading the + configuration file. Behavior has been changed so that these values are + now cleared. + -- 'restrictcid' now properly works on MySQL peers. + -- Only use the default callerid if it has been specified. + -- Asterisk was not sending the same From: line in SIP messages during + certain times. Fixed to make sure it stays the same. This makes some + providers happier, to a working state. + -- Certain circumstances involving a blank callerid caused asterisk to + segmentation fault. + -- There was a problem incorrectly matching codec availablity when global + preferences were different from that of the user. To fix this, + processing of SDP data has been moved to after determining who the call + is coming from. + -- Asterisk would run out of RTP ports while waiting for SUBSCRIBE's to + expire even though an RTP port isn't needed in this case. This has been + fixed by releasing the ports early. + -- chan_zap: + -- During a certain scenario when using flash and '#' transfers you would + hear the other person and the music they were hearing. This has been + fixed. + -- A fix for a compilation issue with gcc4 was added. + -- chan_modem_bestdata: + -- A fix for a compilation issue with gcc4 was added. + -- format_g729: + -- Treat a 10-byte read as an end of file indication instead of an error. + Some G729 encoders like to put 10-bytes at the end to indicate this. + -- res_features: + -- During certain situations when parking a call, both endpoints would get + musiconhold. This has been fixed so the individual who parked the call + will hear the digits and not musiconhold. + -- app_dial: + -- DIALEDPEERNUMBER is now being set, so if you attempted to use it in the + past and failed, it should work now. + -- A callerid change caused many headaches, this has been reversed to the + original 1.0 behavior. + -- A crash caused with the combination of the 'g' option and # transfer was + fixed. + -- app_voicemail: + -- If two people hit the voicemail system at the same time, and were leaving + a message the second message was overwriting the first. This has been + fixed so that each one is distinct and will not overwrite eachother. + -- cdr_tds: + -- If the server you were using was going down, it had the potential to + bring your asterisk server down with it. Extra stuff has been added so + as to bring in more error/connection checking. + -- cdr_pgsql: + -- This will now attempt to reconnect after a connection problem. + -- IAXY firmware: + -- This has been updated to version 23. It includes a fix for lost + registrations. + -- internals + -- Behavior was changed for 'show codec <number>' to make it more intuitive. + -- DNS failures and asterisk do not get along too well, this is not totally + the case anymore. + -- Asterisk will now handle DNS failures at startup more gracefully, and + won't crash and burn + -- Choosing to append to a wave file would render the outputted wave file + corrupt. Appending now works again. + -- If you failed to define certain keys, asterisk had the potential to crash + when seeing if you had used them. + -- Attempting to use such things as ${EXTEN:-1} gave a wrong return value. + However, this was never a documented feature... + +Asterisk 1.0.5 + + -- chan_zap + -- fix a callerid bug introduced in 1.0.4 + -- app_queue + -- fix some penalty behavior + +Asterisk 1.0.4 + + -- general + -- fix memory leak evident with extensive use of variables + -- update IAXy firmware to version 22 + -- enable some special write protection + -- enable outbound DTMF + -- fix seg fault with incorrect usage of SetVar + -- other minor fixes including typos and doc updates + -- chan_sip + -- fix codecs to not be case sensitive + -- Re-use auth credentials + -- fix MWI when using type=friend + -- fix global NAT option + -- chan_agent / chan_local + -- fix incorrect use count + -- chan_zap + -- Allow CID rings to be configured in zapata.conf + -- no more patching needed for UK CID + -- app_macro + -- allow Macros to exit with '*' or '#' like regular extension processing + -- app_voicemail + -- don't allow '#' as a password + -- add option to save voicemail before going to the operator + -- fix global operator=yes + -- app_read + -- return 0 instead of -1 if user enters nothing + -- res_agi + -- don't exit AGI when file not found to stream + -- send script parameter when using FastAGI + +Asterisk 1.0.3 + + -- chan_zap + -- fix seg fault when doing *0 to flash a trunk + -- rtp + -- seg fault fix + -- chan_sip + -- fix to prevent seg fault when attempting a transfer + -- fix bug with supervised transfers + -- fix codec preferences + -- chan_h323 + -- fix compilation problem + -- chan_iax2 + -- avoid a deadlock related to a static config of a BUNCH of peers + -- cdr_pgsql + -- fix memory leak when reading config + -- Numerous other minor bug fixes + +Asterisk 1.0.2 + + -- Major bugfix release + Asterisk 1.0.1 + -- Added AGI over TCP support -- Add ability to purge callers from queue if no agents are logged in -- Fix inband PRI indication detection @@ -23,6 +288,7 @@ Asterisk 1.0.1 -- Fixed seg fault for ast_control_streamfile -- Make pick-up extension configurable via features.conf -- Numerous other bug fixes + Asterisk 1.0.0 -- Use Q.931 standard cause codes for asterisk cause codes -- Bug fixes from the bug tracker |