]>
git.ipfire.org Git - thirdparty/asterisk.git/log
Russell Bryant [Tue, 27 Mar 2007 18:05:46 +0000 (18:05 +0000)]
Merged revisions 59258 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) | 4 lines
Fix the use of the "sourceaddress" option when "bindaddr" is set to 0.0.0.0
instead of having each interface explicitly listed.
(issue #7874, patch by stevens)
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Russell Bryant [Tue, 27 Mar 2007 16:20:53 +0000 (16:20 +0000)]
Convert the RTPQOS function to just be additional parameter of the CHANNEL
function. This way, it will be possible for other RTP based channel drivers
to expose this information in the future.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59256
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Christian Richter [Tue, 27 Mar 2007 15:00:10 +0000 (15:00 +0000)]
Merged revisions 59252 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27 Mär 2007) | 1 line
fixed #9355
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Tilghman Lesher [Mon, 26 Mar 2007 21:45:44 +0000 (21:45 +0000)]
Oops, this should be case insensitive
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59230
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Steve Murphy [Mon, 26 Mar 2007 21:41:32 +0000 (21:41 +0000)]
fix for 9373 (duplicate context in AEL crashes asterisk). I turned a duplicate context from a WARNING to an ERROR. Now you get a module load failure, and asterisk just exits. That's better than a crash, right\?
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59228
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Tilghman Lesher [Mon, 26 Mar 2007 21:37:41 +0000 (21:37 +0000)]
Change this to a single dp function to make oej happy.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59227
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Steve Murphy [Mon, 26 Mar 2007 20:06:12 +0000 (20:06 +0000)]
Fix for 9257; by eliminating the globals in main/config.c, we make it thread-safe, which is a minimum requirement.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59225
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Joshua Colp [Mon, 26 Mar 2007 19:34:14 +0000 (19:34 +0000)]
Add ability to specify no timeout. This means as soon as the prompt is done playing it moves on to the next priority.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59223
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Russell Bryant [Mon, 26 Mar 2007 18:33:50 +0000 (18:33 +0000)]
Somehow the code for building the email for voicemail got out of sync. This
change makes a few tweaks to get 1.4 in sync with trunk.
(issue #9301)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59217
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Russell Bryant [Mon, 26 Mar 2007 18:28:29 +0000 (18:28 +0000)]
Fix some codec negotiation problems when CallerID support is not enabled in SLA.
(issue #9308, reported by twilson)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59215
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Joshua Colp [Mon, 26 Mar 2007 18:13:06 +0000 (18:13 +0000)]
Make SpeechBackground obey the digit timeout value.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59213
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Russell Bryant [Mon, 26 Mar 2007 17:53:07 +0000 (17:53 +0000)]
Rename the new dialplan functions to match the variable name
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59209
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Russell Bryant [Mon, 26 Mar 2007 17:45:55 +0000 (17:45 +0000)]
The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some
because they get set in sip_hangup. So, there are common situations where
the variables will not be available in the dialplan at all. So, this patch
provides an alternate method for getting to this information by introducing
AUDIORTPQOS and VIDEORTPQOS dialplan functions.
(issue #9370, patch by Corydon76, with some testing by blitzrage)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59207
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Steve Murphy [Mon, 26 Mar 2007 17:38:29 +0000 (17:38 +0000)]
A fix for the flex input files, DONT_COMPILE, and STANDALONE_AEL
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59206
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Nadi Sarrar [Mon, 26 Mar 2007 15:25:53 +0000 (15:25 +0000)]
* mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, make chan_misdn use it.
* add a check for linux/mISDNdsp.h to configure.ac and update the autogenerated files: 'configure', 'autoconfig.h.in'
(the 'configure' script was not in sync with the latest configure.ac, so the diff is a bit bigger than expected).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59202
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Joshua Colp [Mon, 26 Mar 2007 15:16:29 +0000 (15:16 +0000)]
Have ast_copy_string magically appear in the aelparse binary! DONT_OPTIMIZE should now work once again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59200
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Joshua Colp [Sat, 24 Mar 2007 01:39:44 +0000 (01:39 +0000)]
Merged revisions 59194 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2 lines
Only try to handle a response if it has a response code. (ASA-2007-011)
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Steve Murphy [Fri, 23 Mar 2007 16:11:15 +0000 (16:11 +0000)]
blocking out the fix in 59187... already incorporated here
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59189
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Steve Murphy [Fri, 23 Mar 2007 16:09:01 +0000 (16:09 +0000)]
Merged revisions 59186 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59186 | murf | 2007-03-23 09:57:26 -0600 (Fri, 23 Mar 2007) | 1 line
Added a few words in the Macro doc strings about the behavior of macros with hangups (et al.), as per 9337
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Kevin P. Fleming [Thu, 22 Mar 2007 23:40:01 +0000 (23:40 +0000)]
don't allow string input to overrun the buffer to hold it (ASA-2007-010)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59182
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Kevin P. Fleming [Thu, 22 Mar 2007 23:34:22 +0000 (23:34 +0000)]
remove variables that are no longer used (--enable-dev-mode is good, developers should be using it)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59180
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Steve Murphy [Thu, 22 Mar 2007 14:40:53 +0000 (14:40 +0000)]
The stuff in utils was compiling with -O6 even if DONT_OPTIMIZE is set in menuconfig. Added the include to fix that
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59145
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Joshua Colp [Wed, 21 Mar 2007 18:08:57 +0000 (18:08 +0000)]
Add svg mimetype for pari.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59089
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Joshua Colp [Wed, 21 Mar 2007 18:04:58 +0000 (18:04 +0000)]
Merged revisions 59086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59086 | file | 2007-03-21 14:03:20 -0400 (Wed, 21 Mar 2007) | 2 lines
Indicate the filename changed when it is changed. (issue #9311 reported by jsmith)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59087
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Joshua Colp [Wed, 21 Mar 2007 03:25:48 +0000 (03:25 +0000)]
Until we can do media level parsing for sendrecv/etc just use the first value found. This crept up when a phone was offered audio+video and returned an inactive video stream. chan_sip thought the phone said to put the person on hold but that was totally wrong. (issue #9319 reported by benbrown)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59081
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Tilghman Lesher [Tue, 20 Mar 2007 21:04:52 +0000 (21:04 +0000)]
Fix defines for inline stack backtraces (only used by developers anyway)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59078
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Joshua Colp [Tue, 20 Mar 2007 20:42:46 +0000 (20:42 +0000)]
Copy len variable as well, should fix remaining IAX2 DTMF issues.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59076
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Steve Murphy [Tue, 20 Mar 2007 17:48:17 +0000 (17:48 +0000)]
Ooops. Sorry, messed up app_stack. This should return it to its previous, untouched, state.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59070
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Steve Murphy [Tue, 20 Mar 2007 17:43:02 +0000 (17:43 +0000)]
The fix for the AEL <<security hole>> (bug 9316) is here...
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59069
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Christian Richter [Tue, 20 Mar 2007 13:16:06 +0000 (13:16 +0000)]
Merged revisions 58849-58850,59062-59063 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) | 1 line
added method standard_dec for dialing out on groups, to avoid conflicts, which caused issues with some ISDN providers
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r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13 Mär 2007) | 1 line
fixed the crypt_keys stuff
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r59062 | crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
avoid sending a disconnect when we already received one.
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r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) | 1 line
modified a loglevel
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Tilghman Lesher [Mon, 19 Mar 2007 22:29:56 +0000 (22:29 +0000)]
Oops, this should have been a %d all along
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59049
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Joshua Colp [Mon, 19 Mar 2007 15:52:28 +0000 (15:52 +0000)]
Fix typo in help for CDR function. (issue #9295 reported by ajohnson)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59042
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Tilghman Lesher [Mon, 19 Mar 2007 15:42:26 +0000 (15:42 +0000)]
Fix unescaped semicolon (reported via -dev list)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59040
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Olle Johansson [Sun, 18 Mar 2007 20:37:06 +0000 (20:37 +0000)]
Issue #9313, Asterisk crash on SIP return code 0 (reported by qwerty1979) (ASA-2007-011)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59037
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BJ Weschke [Sun, 18 Mar 2007 16:36:44 +0000 (16:36 +0000)]
Don't return a non-zero return code if the profile doesn't exist, to match what the documentation says it already does. (#9307 Reported by kkiely)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59035
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Joshua Colp [Fri, 16 Mar 2007 16:12:28 +0000 (16:12 +0000)]
Wait for the async thread to exit when hanging up all of the paged phones under all circumstances. (issue #9181 reported by PhilSmith)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58992
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Russell Bryant [Fri, 16 Mar 2007 01:42:37 +0000 (01:42 +0000)]
fix a couple SLA documentation references
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58957
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Russell Bryant [Fri, 16 Mar 2007 01:41:00 +0000 (01:41 +0000)]
Making these documentation changes in the 1.4 branch upset various people, so
these chanes will only be done in the trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58955
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Russell Bryant [Fri, 16 Mar 2007 01:12:40 +0000 (01:12 +0000)]
Add the --pdf option to the usage of rubber in prep_tarball
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58953
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Russell Bryant [Thu, 15 Mar 2007 23:53:26 +0000 (23:53 +0000)]
Add configure script checking for GTK2 and some additional Makefile targets
to support gmenuselect
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58947
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Tilghman Lesher [Thu, 15 Mar 2007 23:52:48 +0000 (23:52 +0000)]
Refashion dump command to match common syntax and update the resulting appdocs TeX file
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58946
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Russell Bryant [Thu, 15 Mar 2007 23:24:09 +0000 (23:24 +0000)]
add a link to the rubber homepage
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58941
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Tilghman Lesher [Thu, 15 Mar 2007 23:11:33 +0000 (23:11 +0000)]
Expand deprecation warnings from simply warning on use to the builtin documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58939
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Russell Bryant [Thu, 15 Mar 2007 22:51:29 +0000 (22:51 +0000)]
Add Asterisk version information to the generated PDF
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58937
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Russell Bryant [Thu, 15 Mar 2007 22:35:52 +0000 (22:35 +0000)]
have prep_tarball attempt to build asterisk.pdf
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58935
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Tilghman Lesher [Thu, 15 Mar 2007 22:32:33 +0000 (22:32 +0000)]
Function works fine, but the documentation is backwards.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58933
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Russell Bryant [Thu, 15 Mar 2007 22:25:12 +0000 (22:25 +0000)]
Merge changes from svn/asterisk/team/russell/LaTeX_docs.
* Convert most of the doc directory into a single LaTeX formatted document
so that we can generate a PDF, HTML, or other formats from this
information.
* Add a CLI command to dump the application documentation into LaTeX format
which will only be include if the configure script is run with
--enable-dev-mode.
* The PDF turned out to be close to 1 MB, so it is not included. However, you
can simply run "make asterisk.pdf" to generate it yourself. We may include
it in release tarballs or have automatically generated ones on the web site,
but that has yet to be decided.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58931
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Joshua Colp [Thu, 15 Mar 2007 18:13:21 +0000 (18:13 +0000)]
Don't assume that the pvt structure will still exist after calling schedule_delivery as it may not. (issue #9278 reported by fmachado)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58923
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Russell Bryant [Wed, 14 Mar 2007 19:18:08 +0000 (19:18 +0000)]
Some people like to put "limitonpeer" instead of "limitonpeers" in their
configuration. While we're at it, support "limitonpeerz" and
"limitonpeerssssss". (inspired by issue #9172)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58906
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Russell Bryant [Wed, 14 Mar 2007 17:04:38 +0000 (17:04 +0000)]
Add a more basic example setup to the examples section
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58902
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Russell Bryant [Wed, 14 Mar 2007 16:40:22 +0000 (16:40 +0000)]
Merged revisions 58896 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) | 3 lines
Add a note to the security file that the Asterisk CLI and log files may contain
sensitive information, and that people should keep this in mind.
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Russell Bryant [Wed, 14 Mar 2007 16:33:01 +0000 (16:33 +0000)]
By default, don't attempt to do any CallerID handling at all with SLA because
it is known to not work properly in some situations. However, add an option to
enable it for those that would like to use it anyway.
The short story behind this is that to properly handle CallerID with SLA, we
need the ability to change the CallerID on an existing call, and we are not
ready to handle that.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58894
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Tilghman Lesher [Wed, 14 Mar 2007 01:47:08 +0000 (01:47 +0000)]
Issue 9162 - pbx_substitute_variables_helper assumes the buffer is initialized
to all zeroes. This fixes a case where it wasn't.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58880
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Russell Bryant [Tue, 13 Mar 2007 23:19:51 +0000 (23:19 +0000)]
Ensure that the blinky lights show that the trunk stopped ringing when the
trunk hangs up before a station has answered it.
(issue #9234, reported by francesco_r)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58872
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Russell Bryant [Tue, 13 Mar 2007 23:11:08 +0000 (23:11 +0000)]
fix the reference to the SLA documentation
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58870
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Olle Johansson [Tue, 13 Mar 2007 11:49:35 +0000 (11:49 +0000)]
Merged revisions 58847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 lines
Issue #9229 - No port in request URI on register to non default SIP ports (neelakantan)
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Olle Johansson [Tue, 13 Mar 2007 10:03:03 +0000 (10:03 +0000)]
Don't hangup the call on OK or errors on MESSAGE and INFO
inside of a dialog (like video update requests).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58845
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Olle Johansson [Tue, 13 Mar 2007 09:12:16 +0000 (09:12 +0000)]
Issue #9251 - Clear From URI from user attributes (tgrman)
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Joshua Colp [Mon, 12 Mar 2007 16:52:14 +0000 (16:52 +0000)]
Blocked revisions 58832 via svnmerge
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r58832 | file | 2007-03-12 12:49:49 -0400 (Mon, 12 Mar 2007) | 2 lines
We can't use the assembler version of fetchadd_int under Intel Macs. (issue #9254 reported by darrell budic)
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Christian Richter [Mon, 12 Mar 2007 13:08:06 +0000 (13:08 +0000)]
Merged revisions 57034,57523,57753,58558 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) | 1 line
fixed bugs.digium.com bugs: #9157 and bugs.beronet.com bugs: #302, #303, #304
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r57523 | crichter | 2007-03-02 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line
fixed typo
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r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) | 1 line
fixed another place where the out_cause was hardcoded to 16
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r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09 Mar 2007) | 1 line
we can free channel 31 as well, since we can occupy it
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Christian Richter [Mon, 12 Mar 2007 12:43:24 +0000 (12:43 +0000)]
added UU transceiving and corect handling for rdnis
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Joshua Colp [Mon, 12 Mar 2007 01:21:12 +0000 (01:21 +0000)]
Allow RFC2833 compensation to compensate for even stupider implementations by queueing up the end frame at the start, not the actual end. (issue #8963 reported by AndrewZ)
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Joshua Colp [Mon, 12 Mar 2007 00:51:16 +0000 (00:51 +0000)]
Add matchexterniplocally setting which only substitutes your externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska)
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Russell Bryant [Sat, 10 Mar 2007 18:11:11 +0000 (18:11 +0000)]
Fix a few more places in chan_iax2 where the ast_frame used for receiving a
frame was not properly initialized.
- Interpolating a frame when the jitterbuffer is in use
- decrypting a frame when IAX2 encryption is on
- frames in an IAX2 trunk
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Russell Bryant [Sat, 10 Mar 2007 03:58:27 +0000 (03:58 +0000)]
Make the compiler happy and initialize a variable.
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Russell Bryant [Fri, 9 Mar 2007 23:59:10 +0000 (23:59 +0000)]
Merge some updates to the SLA documentation. I plan to keep working on this
to explain all of the expected behavior with call handling, configuration
details for specific phones, and other things. However, I got tired of doing
it in plain text, so I switched to using LaTeX. I have included the PDF version.
I haven't been able to get a nice looking plain text version out of it yet, but
I'm not terribly concerned since this is supposed to be more of the manual,
while the plain text sample configuration file is the reference.
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Joshua Colp [Fri, 9 Mar 2007 21:08:19 +0000 (21:08 +0000)]
Fix spelling of unavailable in voicemail documentation. (issue #9248 reported by tensai)
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Joshua Colp [Fri, 9 Mar 2007 20:49:47 +0000 (20:49 +0000)]
Merged revisions 58579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 lines
If we are unable to lookup the host in a c line we have to abort, otherwise the previous data is gone and we will (potentially) have no data when all is said and done.
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Russell Bryant [Thu, 8 Mar 2007 22:15:15 +0000 (22:15 +0000)]
Hang up the channel that put the call on hold in the event processing thread to
avoid a race condition. Also, if the station originated the call that it is
putting on hold, don't hang up the trunk if it was the only station on the call
and it is hanging up due to hold and not a normal hangup.
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Russell Bryant [Thu, 8 Mar 2007 22:06:54 +0000 (22:06 +0000)]
Add a missing break statement so that handling the above event does not
incorrectly destroy the channel. (issue #9242, andrew)
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Tilghman Lesher [Thu, 8 Mar 2007 21:33:03 +0000 (21:33 +0000)]
Fix segfault (Issue 9236)
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Russell Bryant [Thu, 8 Mar 2007 20:54:56 +0000 (20:54 +0000)]
Refactor hold handling a bit so that it does not require keeping the call up
when a call is put on hold.
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Joshua Colp [Thu, 8 Mar 2007 18:01:00 +0000 (18:01 +0000)]
Make early SDP seeding even smarter! We have to check codecs in the make_compatible function too. (issue #9221 reported by marcelbarbulescu)
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Joshua Colp [Thu, 8 Mar 2007 16:07:10 +0000 (16:07 +0000)]
Merged revisions 58388 via svnmerge from
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r58388 | file | 2007-03-08 11:04:58 -0500 (Thu, 08 Mar 2007) | 2 lines
Only print out debug message if the definition that makes the variables shows up was actually defined. (issue #9233 reported by serginuez)
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Kevin P. Fleming [Thu, 8 Mar 2007 13:23:46 +0000 (13:23 +0000)]
this change was not needed; fclose() handles closing the file descriptor already
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Kevin P. Fleming [Thu, 8 Mar 2007 13:17:42 +0000 (13:17 +0000)]
fix a compiler warning, and overwriting 'res' value
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Kevin P. Fleming [Thu, 8 Mar 2007 13:17:17 +0000 (13:17 +0000)]
fix two cases where HTTP session file descriptors would not be closed
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Russell Bryant [Thu, 8 Mar 2007 01:01:46 +0000 (01:01 +0000)]
If we receive ZT_EVENT_REMOVED, destroy the specified channel.
(issue #7256, tzafrir)
Also, update the configure script to make sure that we don't try to build
chan_zap if the installed version of zaptel does not include ZT_EVENT_REMOVED.
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Russell Bryant [Wed, 7 Mar 2007 18:19:19 +0000 (18:19 +0000)]
(This bug was reported to me by Kinsey Moore)
Merged revisions 58242 via svnmerge from
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r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | 7 lines
Fix a problem where the Asterisk channel name could be that of the wrong IAX2
user for a call. This is because the first step of choosing this name is to
look for an IAX2 peer that happens to have the same IP/port number that this
call is coming from and assuming that is it. However, this is not always
correct. So, I have made it change this name after authentication happens
since at that point, we have an exact match.
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Joshua Colp [Wed, 7 Mar 2007 17:52:58 +0000 (17:52 +0000)]
Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu)
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Russell Bryant [Wed, 7 Mar 2007 00:27:59 +0000 (00:27 +0000)]
Blocked revisions 58167 via svnmerge
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r58167 | russell | 2007-03-06 18:27:04 -0600 (Tue, 06 Mar 2007) | 2 lines
Fix a misplaced block of code in the 1.2 version of the patch to fix issue #8977
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Russell Bryant [Wed, 7 Mar 2007 00:25:19 +0000 (00:25 +0000)]
Merged revisions 58164 via svnmerge from
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r58164 | russell | 2007-03-06 18:20:13 -0600 (Tue, 06 Mar 2007) | 4 lines
If the channels acquired using the manager Redirect action are not up, then
don't attempt to do anything with them. It could lead to weird behavior,
including crashes. (issue #8977)
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Steve Murphy [Tue, 6 Mar 2007 23:10:14 +0000 (23:10 +0000)]
Merged revisions 58115 via svnmerge from
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r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 line
Fix for 9220: Eyebeam cannot renew subscriptions for presence info. Reason: re-SUBSCRIBE requests don't include Accept headers, which the rfc says are optional (to put it tersely), (it uses MAY), and luckily, the sip_pvt struct has the format info stored, so we simply leave it if the format is set, and the accept header null.
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Russell Bryant [Tue, 6 Mar 2007 23:00:57 +0000 (23:00 +0000)]
Clarify the documentation of the dialout and sendvoicemail options.
(issue #9000, caio1982 and serge-v)
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Olle Johansson [Tue, 6 Mar 2007 20:37:07 +0000 (20:37 +0000)]
Merged revisions 58052 via svnmerge from
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r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 lines
Change error message to proper message
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Russell Bryant [Tue, 6 Mar 2007 18:01:20 +0000 (18:01 +0000)]
Return an error of transmit_response is called without a session.
(issue #9002)
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Joshua Colp [Mon, 5 Mar 2007 19:19:07 +0000 (19:19 +0000)]
Since chan_iax2 does not support reception of DTMF with duration ensure that it is set to 0 on the frame. (issue #8521 reported by gdhgdh)
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Joshua Colp [Mon, 5 Mar 2007 18:39:28 +0000 (18:39 +0000)]
Don't create a listen channel and record the conference unless the option is turned on. (issue #9204 reported by francesco_r)
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Joshua Colp [Mon, 5 Mar 2007 17:52:03 +0000 (17:52 +0000)]
Merged revisions 57869 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57869 | file | 2007-03-05 12:49:18 -0500 (Mon, 05 Mar 2007) | 2 lines
Make create_dirpath use our standard for return values. -1 is failure, 0 is success. (issue #9205 reported by ballares)
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Steve Murphy [Mon, 5 Mar 2007 15:20:17 +0000 (15:20 +0000)]
Merged revisions 57825 via svnmerge from
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r57825 | murf | 2007-03-05 07:53:57 -0700 (Mon, 05 Mar 2007) | 1 line
Fixed a typo introduced via 9156 (either the gotos or their doc strings are wrong)
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Joshua Colp [Mon, 5 Mar 2007 04:19:53 +0000 (04:19 +0000)]
Don't allow a NULL pointer to reach ast_frdup. (issue #9155 reported by cmaj)
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Joshua Colp [Mon, 5 Mar 2007 03:35:03 +0000 (03:35 +0000)]
Don't reference a potentially NULL pointer. (issue #9199 reported by klolik)
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Joshua Colp [Mon, 5 Mar 2007 03:22:17 +0000 (03:22 +0000)]
Preserve marker bit when P2P bridging. (issue #9198 reported by edgreenberg)
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Steve Murphy [Sat, 3 Mar 2007 15:31:12 +0000 (15:31 +0000)]
Updated the regression tests
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Tilghman Lesher [Sat, 3 Mar 2007 06:45:00 +0000 (06:45 +0000)]
Merged revisions 57648 via svnmerge from
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r57648 | tilghman | 2007-03-03 00:36:55 -0600 (Sat, 03 Mar 2007) | 2 lines
Memory leak of a list, if call recording was abandoned
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Dwayne M. Hubbard [Sat, 3 Mar 2007 00:59:24 +0000 (00:59 +0000)]
submitted patch for Georgian language, issue 9010, submitted by Alexander Shaduri
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Russell Bryant [Sat, 3 Mar 2007 00:02:29 +0000 (00:02 +0000)]
add missing configuration template. Thanks to Lacy Moore on asterisk-users for pointing this out\!
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Russell Bryant [Fri, 2 Mar 2007 23:03:01 +0000 (23:03 +0000)]
Update the check that is used to determine whether zaptel transcoder support
is present. The interface has changed.
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Joshua Colp [Fri, 2 Mar 2007 17:06:52 +0000 (17:06 +0000)]
Merged revisions 57475 via svnmerge from
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r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 lines
If a SIP message comes in and goes to a method handler that requires additional values that may not be present then send back an error.
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Steve Murphy [Fri, 2 Mar 2007 16:55:16 +0000 (16:55 +0000)]
Merged revisions 57458 via svnmerge from
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r57458 | murf | 2007-03-02 09:39:33 -0700 (Fri, 02 Mar 2007) | 1 line
further refinement in wording of goto documentation, as per 9156, goto not proceeding to next instruction
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