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r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines
Merged revisions 328209 via svnmerge from
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r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines
Introduce <support_level> tags in MODULEINFO.
This change introduces MODULEINFO into many modules in Asterisk in order to show
the community support level for those modules. This is used by changes committed
to menuselect by Russell Bryant recently (r917 in menuselect). More information about
the support level types and what they mean is available on the wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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2011-07-14 20:28:54 +00:00
4aca3187a3
Asterisk media architecture conversion - no more format bitfields
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This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal. For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs. Functionally
no change in behavior should be present in this patch. Thanks to twilson
and russell for all the time they spent reviewing these changes.
Review: https://reviewboard.asterisk.org/r/1083/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306010 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-03 16:22:10 +00:00
82c3385315
Merged revisions 284610 via svnmerge from
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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: http://svn.digium.com/svn/asterisk/trunk@284628 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-02 05:27:53 +00:00
9f18f034a5
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: http://svn.digium.com/svn/asterisk/trunk@264335 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-19 20:02:57 +00:00
3bacd4082e
Expand codec bitfield from 32 bits to 64 bits.
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Reviewboard: https://reviewboard.asterisk.org/r/416/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227580 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04 14:05:12 +00:00
230a66da7d
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: http://svn.digium.com/svn/asterisk/trunk@196072 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-21 21:13:09 +00:00
21d93ce816
Swap reversed timevals.
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This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@179254 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-01 23:25:23 +00:00
d7dab67f63
Merged revisions 177383 via svnmerge from
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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
17e96df89a
Move Speech* applications and functions documentation to XML.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@161536 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-06 21:18:51 +00:00
c407ae1a7d
Merged revisions 147517 via svnmerge from
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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
74ecba3091
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146802 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-06 21:09:05 +00:00
0891b8a53c
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135680 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05 16:56:11 +00:00
c1210321e7
- 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: http://svn.digium.com/svn/asterisk/trunk@117802 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-22 16:29:54 +00:00
d1cc29c9c1
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: http://svn.digium.com/svn/asterisk/trunk@115076 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-01 23:06:23 +00:00
712c72b0df
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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114904 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-30 19:21:04 +00:00
48f62970c3
Whitespace changes only
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2008-04-25 20:20:10 +00:00
84aa522629
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
e2f5175dcd
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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@101082 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-30 00:04:17 +00:00
9cf442d7f7
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19 18:52:04 +00:00
883346d64a
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.).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16 20:04:58 +00:00
92ac6820ee
"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
cfe23d813b
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
4d6bda5445
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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
356721a45c
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
5c0772ce7d
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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
d17ff1ea42
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75200 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16 14:39:29 +00:00
9e24ed5ccf
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75183 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16 13:35:20 +00:00
f9980e7f7c
Use the linkedlists.h AST_LIST_NEXT macro for modifying the list of results.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@74616 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-11 17:34:30 +00:00
13e34e0fa9
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@74570 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-11 16:19:00 +00:00
87786050a2
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
0a0ae2c364
Merged revisions 69558 via svnmerge from
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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: http://svn.digium.com/svn/asterisk/trunk@69559 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-15 19:25:11 +00:00
7a0fe5c93f
Convert uses of strdup() to ast_strdup()
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(issue #9983 , eliel)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@69436 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-14 23:01:01 +00:00
eb5d461ed4
Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67864 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-06 21:20:11 +00:00
3ef46b6a1c
Merged revisions 61651 via svnmerge from
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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
5f1367dd74
Merged revisions 60361 via svnmerge from
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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
38bb3d1841
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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
01e2d46c58
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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
072a1a4133
Merged revisions 59213 via svnmerge from
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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
9c34ba77f7
Merged revisions 57053 via svnmerge from
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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
5f1271f728
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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
9a17fb8b1e
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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
e617d4e0d4
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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
62f5251e78
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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
1dc669e571
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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
ec3737e835
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
6c89612673
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44435 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-05 01:40:06 +00:00
03ecb2c2d7
Documentation updates (thanks Shaun for the speechrec.txt one!)
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2006-08-24 15:44:24 +00:00
8b0c007ad9
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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-21 02:11:39 +00:00
1e12d4e1df
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37881 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-18 16:22:26 +00:00
556cad8ac4
don't leak a frame when breaking out of the loop on a timeout
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@33448 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-11 14:52:04 +00:00
lmadsen