8190e96fad
Merged revisions 284610 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines
When optional_api is non-optional, force dependent modules to be loaded.
(closes issue #17707 )
Reported by: ira
Patches:
20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
Review: https://reviewboard.asterisk.org/r/876/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-02 05:27:53 +00:00
d38c6459f5
Merged revisions 264334 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19 20:02:57 +00:00
d8e0c58437
Expand codec bitfield from 32 bits to 64 bits.
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Reviewboard: https://reviewboard.asterisk.org/r/416/
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2009-11-04 14:05:12 +00:00
e6b2e9a750
Const-ify the world (or at least a good part of it)
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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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21 21:13:09 +00:00
932cef55fd
Swap reversed timevals.
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This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ!
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2009-03-01 23:25:23 +00:00
2e2494591e
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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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2009-02-19 16:38:41 +00:00
206fe71680
Move Speech* applications and functions documentation to XML.
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2008-12-06 21:18:51 +00:00
fddf5f914a
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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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2008-10-08 14:53:51 +00:00
63b165dbb9
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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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2008-10-06 21:09:05 +00:00
7df8b8b848
make datastore creation and destruction a generic API since it is not really channel related, and add the ability to add/find/remove datastores to manager sessions
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2008-08-05 16:56:11 +00:00
f1e9371da8
- revert change to ast_queue_hangup and create ast_queue_hangup_with_cause
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-22 16:29:54 +00:00
b5a127daac
Modify TIMEOUT() to be accurate down to the millisecond.
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(closes issue #10540 )
Reported by: spendergrass
Patches:
20080417__bug10540.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-01 23:06:23 +00:00
527c601d0c
Lock around variables retrieved, and copy the values, if they stay persistent,
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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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2008-04-30 19:21:04 +00:00
463a5dbd0a
Whitespace changes only
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2008-04-25 20:20:10 +00:00
8718878490
Merged revisions 106552 via svnmerge from
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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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2008-03-07 06:54:47 +00:00
22fae48e3c
Add the 'n' option to SpeechBackground, which has the application not answer the
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channel if it has not already been answered.
(closes SPD-51)
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2008-01-30 00:04:17 +00:00
0595b5e2aa
include "logger.h" and errno.h from asterisk.h - usage shows that they
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were included almost everywhere.
Remove some of the instances.
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2007-11-19 18:52:04 +00:00
fdb7f7ba3d
Start untangling header inclusion in a way that does not affect
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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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2007-11-16 20:04:58 +00:00
5a4867543d
"show application <foo>" changes for clarity.
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(closes issue #11171 , reported and patched by blitzrage)
Many thanks!
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2007-11-06 19:04:45 +00:00
94b1dda1fa
Merged revisions 79334 via svnmerge from
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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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2007-08-13 21:59:15 +00:00
3dac92ecbf
Merged revisions 79207 via svnmerge from
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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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2007-08-13 14:55:17 +00:00
20bbd09de3
Mostly cleanup of documentation to substitute the pipe with the comma, but a few other formatting cleanups, too.
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2007-07-31 01:10:47 +00:00
1c783e8cfb
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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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2007-07-25 22:18:56 +00:00
b8cd949cce
Applications no longer need to call ast_module_user_add and ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application.
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2007-07-16 14:39:29 +00:00
96a646734f
It is no longer required for each module that deals with a channel to call ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it.
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2007-07-16 13:35:20 +00:00
7e015109a2
Use the linkedlists.h AST_LIST_NEXT macro for modifying the list of results.
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2007-07-11 17:34:30 +00:00
f8e605fe25
Allow the native formats of a channel to influence the audio that is going to the engine. The best format will try to be chosen with an ultimate fallback to signed linear if possible.
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2007-07-11 16:19:00 +00:00
68ac7f7633
Merged revisions 71068 via svnmerge from
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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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2007-06-22 15:03:32 +00:00
84e624ff23
Merged revisions 69558 via svnmerge from
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-15 19:25:11 +00:00
b179e2155f
Convert uses of strdup() to ast_strdup()
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(issue #9983 , eliel)
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2007-06-14 23:01:01 +00:00
9d05ff8ed5
Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
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2007-06-06 21:20:11 +00:00
67302c5411
Merged revisions 61651 via svnmerge from
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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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2007-04-13 18:09:29 +00:00
4b618442a0
Merged revisions 60361 via svnmerge from
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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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2007-04-06 01:15:50 +00:00
d6f029b4f7
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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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2007-04-03 19:43:26 +00:00
e2c9d8b43b
Merged revisions 59223 via svnmerge from
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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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2007-03-26 19:35:24 +00:00
722962dec4
Merged revisions 59213 via svnmerge from
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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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2007-03-26 18:14:33 +00:00
902319882c
Merged revisions 57053 via svnmerge from
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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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2007-02-28 17:47:41 +00:00
5030e05632
Merged revisions 55947 via svnmerge from
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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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2007-02-21 20:05:15 +00:00
56095555da
Merged revisions 54714 via svnmerge from
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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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2007-02-16 00:54:10 +00:00
bcb5daf74d
Merged revisions 53601 via svnmerge from
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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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2007-02-08 17:56:53 +00:00
68d39c2b72
Merged revisions 51251 via svnmerge from
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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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2007-01-18 19:19:24 +00:00
4aba5a78d9
Merged revisions 50433 via svnmerge from
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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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2007-01-10 20:26:55 +00:00
cd73a483f1
const-ify some more APIs, and fix rev 49710 from branch-1.4 in a better way here
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2007-01-05 23:32:42 +00:00
3d323f5345
As per ToDo list, I have made it so that Wait(), WaitExten(), Congestion(), 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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2006-10-05 01:40:06 +00:00
5345469008
Documentation updates (thanks Shaun for the speechrec.txt one!)
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2006-08-24 15:44:24 +00:00
0a27d8bfe5
merge new_loader_completion branch, including (at least):
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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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2006-08-21 02:11:39 +00:00
457ec51f8d
Expand speech API so that the developer can interact with the engine more directly and use specific functions of the connector even if a generic API call is not available
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2006-07-18 16:22:26 +00:00
558d3924ac
don't leak a frame when breaking out of the loop on a timeout
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2006-06-11 14:52:04 +00:00
472c1ca282
simplify autoconfig include mechanism (make tholo happy he can use lint again :-)
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2006-06-07 18:54:56 +00:00
18606233da
fix various typos and other bits (from Ian Kinner)
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2006-05-30 16:01:50 +00:00
Tilghman Lesher
Matthew Nicholson
Kevin P. Fleming
Mark Michelson
Joshua Colp
Eliel C. Sardanons
Michiel van Baak
Russell Bryant
Luigi Rizzo
Jason Parker
Steve Murphy