Merge ChangeLog from the v1-0 branch and begin a major feature addition list

for 1.2.  I know this list is very incomplete so anyone that would like to help
add stuff, please contact me.  (No, 1.0.10 hasn't been released.  That is going
to come out with 1.2).


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6738 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 283 insertions, 17 deletions

diff --git a/CHANGES b/CHANGES
index 13cafe4d7..dee603de1 100755
--- a/CHANGES
+++ b/CHANGES
@@ -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