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fix callerid matching in extensions.conf

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formatting fixes for the ChangeLog


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/v1-0@6014 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.0
Russell Bryant 21 years ago
parent cf238fa10d
commit 5fe5bf08b2
2 changed files with 100 additions and 89 deletions
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176
CHANGES
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@ -1,48 +1,50 @@
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.
-- chan_mgcp
-- *70 is used to disable call waiting. Call waiting will now be re-enabled
on hangup.
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.
-- 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.
-- Add support for feautres of 2nd gen hardware
-- 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
@ -52,15 +54,17 @@ Asterisk 1.0.8
-- 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
@ -70,79 +74,91 @@ Asterisk 1.0.8
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

13
pbx/pbx_config.c
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@ -1687,15 +1687,10 @@ static int pbx_load_module(void)
else
data = "";
}
pbx_substitute_variables_helper(NULL, ext, realext, sizeof(realext)-1);
cidmatch = strchr(ext, '/');
if (cidmatch) {
*cidmatch = '\0';
cidmatch++;
}
stringp=ext;
strsep(&stringp, "/");
pbx_substitute_variables_helper(NULL, ext, realext, sizeof(realext)-1);
stringp = realext;
ext = strsep(&stringp, "/");
cidmatch = stringp;
if (!data)
data="";
while(*appl && (*appl < 33)) appl++;

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