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https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines
Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent.
(closes issue #16903)
Reported by: Nick_Lewis
Patches:
pbx.c-specificity.patch uploaded by Nick Lewis (license 657)
Tested by: Nick_Lewis
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r264334 | mnicholson | 2010-05-19 15:01:38 -0500 (Wed, 19 May 2010) | 5 lines
Set quieted flag when receiving a dtmf tone during playback in speechbackground.
(closes issue #16966)
Reported by: asackheim
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Reviewboard: https://reviewboard.asterisk.org/r/416/
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This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/
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This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ!
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3 lines
If we are able to create a speech structure unset the ERROR variable in case it was previously set.
(issue #LUMENVOX-13)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) | 2 lines
If we receive DTMF make sure that the state of the speech structure goes back to being not ready. (issue #LUMENVOX-8)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008) | 8 lines
Dialplan functions should not actually return 0, unless they have modified the
workspace. To signal an error (and no change to the workspace), -1 should be
returned instead.
(closes issue #13340)
Reported by: kryptolus
Patches:
20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14)
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channel related, and add the ability to add/find/remove datastores to manager sessions
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135680 f38db490-d61c-443f-a65b-d21fe96a405b
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- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
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(closes issue #10540)
Reported by: spendergrass
Patches:
20080417__bug10540.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
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since another thread could remove them.
(closes issue #12541)
Reported by: snuffy
Patches:
bug_12156_apps.diff uploaded by snuffy (license 35)
Several additional changes by me
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008) | 6 lines
Safely use the strncat() function.
(closes issue #11958)
Reported by: norman
Patches:
20080209__bug11958.diff.txt uploaded by Corydon76 (license 14)
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channel if it has not already been answered.
(closes SPD-51)
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were included almost everywhere.
Remove some of the instances.
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build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
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(closes issue #11171, reported and patched by blitzrage)
Many thanks!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89044 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r79334 | file | 2007-08-13 18:57:20 -0300 (Mon, 13 Aug 2007) | 2 lines
Instead of accepting a single DTMF character accept a full string.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r79207 | file | 2007-08-13 11:51:09 -0300 (Mon, 13 Aug 2007) | 2 lines
Add an API call to allow the engine to know that DTMF was received.
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few other formatting cleanups, too.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77808 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77176 | file | 2007-07-25 19:16:10 -0300 (Wed, 25 Jul 2007) | 4 lines
(closes issue #10303)
Reported by: jtodd
Add SPEECH_DTMF_TERMINATOR variable so the user can specify the digit to terminate a DTMF string with. If none is specified then no terminator will be used.
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ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application.
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ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it.
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to the engine. The best format will try to be chosen with an ultimate fallback to signed linear if possible.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r71068 | qwell | 2007-06-22 10:00:30 -0500 (Fri, 22 Jun 2007) | 12 lines
Merged revisions 71065 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4 lines
Fix a few silly usages of ast_playstream() - it only ever returns 0...
Issue 10035
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r69558 | file | 2007-06-15 15:23:45 -0400 (Fri, 15 Jun 2007) | 2 lines
Add support for setting the maximum length of acceptable DTMF in SpeechBackground.
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(issue #9983, eliel)
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guidelines changes
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61651 | file | 2007-04-13 14:08:02 -0400 (Fri, 13 Apr 2007) | 2 lines
Do not bother looking for a result if none are present.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r60361 | file | 2007-04-05 22:14:00 -0300 (Thu, 05 Apr 2007) | 2 lines
Add support for returning different types of results (ie: NBest).
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r59963 | file | 2007-04-03 15:40:59 -0400 (Tue, 03 Apr 2007) | 2 lines
Don't clash when a person both speaks and uses DTMF.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r59223 | file | 2007-03-26 16:34:14 -0300 (Mon, 26 Mar 2007) | 2 lines
Add ability to specify no timeout. This means as soon as the prompt is done playing it moves on to the next priority.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r59213 | file | 2007-03-26 14:13:06 -0400 (Mon, 26 Mar 2007) | 2 lines
Make SpeechBackground obey the digit timeout value.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r57053 | file | 2007-02-28 12:45:50 -0500 (Wed, 28 Feb 2007) | 2 lines
Better handle timeouts when the individual speaks after everything has been played but before the timeout ends.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55947 | file | 2007-02-21 15:03:38 -0500 (Wed, 21 Feb 2007) | 2 lines
Only start playing the next file if we have not been quieted.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54714 | file | 2007-02-15 19:48:48 -0500 (Thu, 15 Feb 2007) | 2 lines
Don't let dtmf leak over into the engine and let it skew the results... also give DTMF results priority. (issue #9014 reported by surftek)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53601 | file | 2007-02-08 12:54:32 -0500 (Thu, 08 Feb 2007) | 2 lines
Fix timeout issue when utterance is longer then timeout itself.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r51251 | file | 2007-01-18 14:17:34 -0500 (Thu, 18 Jan 2007) | 2 lines
Only start timeout once we reach the end of the files to play back.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r50433 | file | 2007-01-10 15:25:44 -0500 (Wed, 10 Jan 2007) | 2 lines
Merge speech-multi branch which adds support for joining multiple sound files together to be played one after another in SpeechBackground.
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Busy(), Read(), WaitForRing(), will now either actually handle a floating point argument as advertised, or has been upgraded to accept a floating point [timeout] arg.
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- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
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directly and use specific functions of the connector even if a generic API call is not available
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@33448 f38db490-d61c-443f-a65b-d21fe96a405b
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again :-)
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