[Asterisk-cvs] asterisk ChangeLog,1.74.2.53.2.1,1.74.2.53.2.2
russell at lists.digium.com
russell at lists.digium.com
Wed Jun 29 17:01:27 CDT 2005
Update of /usr/cvsroot/asterisk
In directory mongoose.digium.com:/tmp/cvs-serv12705
Modified Files:
Tag: v1-0-8-fixed
ChangeLog
Log Message:
formatting fixes
Index: ChangeLog
===================================================================
RCS file: /usr/cvsroot/asterisk/ChangeLog,v
retrieving revision 1.74.2.53.2.1
retrieving revision 1.74.2.53.2.2
diff -u -d -r1.74.2.53.2.1 -r1.74.2.53.2.2
--- ChangeLog 29 Jun 2005 20:27:36 -0000 1.74.2.53.2.1
+++ ChangeLog 29 Jun 2005 21:01:20 -0000 1.74.2.53.2.2
@@ -1,8 +1,7 @@
- NOTE: Corrections or additions to the ChangeLog may be submitted
- 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.
+ 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 1.0.9
@@ -11,37 +10,41 @@
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.
+ -- 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.
+ -- 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
+ -- 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
+ -- 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"
+ -- 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.
+ -- 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
@@ -51,15 +54,17 @@
-- 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
+ -- 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
+ -- 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
+ -- 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
+ -- 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
@@ -69,79 +74,91 @@
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.
+ -- 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.
+ -- 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.
+ -- 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.
+ -- 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.
+ -- 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.
+ -- 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.
+ -- 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.
+ -- 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.
+ -- 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.
+ -- 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.
+ -- 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.
+ -- 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. (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...
+ -- 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
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