changelog for 1.0.6, thanks file!

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-0@5072 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES

@@ -1,14 +1,66 @@
  NOTE: Corrections or additions to the ChangeLog may be submitted 
-       to http://bugs.digium.com.  A complete listing of changes
+       to http://bugs.digium.com.  Documentation and formatting
+       fixes are not listed here.  A complete listing of changes
        is available through the Asterisk-CVS mailing list hosted
        at http://lists.digium.com.
 
+Asterisk 1.0.6
+
+ -- 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.
+ -- 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.
+ -- 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.
+ -- 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.
+ -- internals
+    -- Behavior was changed for 'show codec <number>' to make it more intuitive. (kshumard)
+    -- 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
@@ -37,7 +89,9 @@ Asterisk 1.0.4
  -- 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
@@ -53,9 +107,13 @@ Asterisk 1.0.3
  -- 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
@@ -63,6 +121,7 @@ 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