Joshua Colp
fddf5f914a
Merged revisions 147517 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147518 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-08 14:53:51 +00:00
Tilghman Lesher
63b165dbb9
Merged revisions 146799 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-06 21:09:05 +00:00
Kevin P. Fleming
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-05 16:56:11 +00:00
Michiel van Baak
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
Tilghman Lesher
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
Tilghman Lesher
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-30 19:21:04 +00:00
Tilghman Lesher
463a5dbd0a
Whitespace changes only
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114667 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-25 20:20:10 +00:00
Tilghman Lesher
8718878490
Merged revisions 106552 via svnmerge from
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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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2008-03-07 06:54:47 +00:00
Russell Bryant
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-30 00:04:17 +00:00
Luigi Rizzo
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19 18:52:04 +00:00
Luigi Rizzo
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.).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16 20:04:58 +00:00
Mark Michelson
5a4867543d
"show application <foo>" changes for clarity.
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(closes issue #11171 , reported and patched by blitzrage)
Many thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06 19:04:45 +00:00
Joshua Colp
94b1dda1fa
Merged revisions 79334 via svnmerge from
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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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2007-08-13 21:59:15 +00:00
Joshua Colp
3dac92ecbf
Merged revisions 79207 via svnmerge from
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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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2007-08-13 14:55:17 +00:00
Tilghman Lesher
20bbd09de3
Mostly cleanup of documentation to substitute the pipe with the comma, but a few other formatting cleanups, too.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-31 01:10:47 +00:00
Joshua Colp
1c783e8cfb
Merged revisions 77176 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-25 22:18:56 +00:00
Joshua Colp
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-16 14:39:29 +00:00
Joshua Colp
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-16 13:35:20 +00:00
Joshua Colp
7e015109a2
Use the linkedlists.h AST_LIST_NEXT macro for modifying the list of results.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-11 17:34:30 +00:00
Joshua Colp
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-11 16:19:00 +00:00
Jason Parker
68ac7f7633
Merged revisions 71068 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-22 15:03:32 +00:00
Joshua Colp
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
Russell Bryant
b179e2155f
Convert uses of strdup() to ast_strdup()
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(issue #9983 , eliel)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-14 23:01:01 +00:00
Tilghman Lesher
9d05ff8ed5
Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-06 21:20:11 +00:00
Joshua Colp
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
Joshua Colp
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
Joshua Colp
d6f029b4f7
Merged revisions 59963 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59969 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-03 19:43:26 +00:00
Joshua Colp
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-26 19:35:24 +00:00
Joshua Colp
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59214 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-26 18:14:33 +00:00
Joshua Colp
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57054 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-28 17:47:41 +00:00
Joshua Colp
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21 20:05:15 +00:00
Joshua Colp
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16 00:54:10 +00:00
Joshua Colp
bcb5daf74d
Merged revisions 53601 via svnmerge from
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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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2007-02-08 17:56:53 +00:00
Joshua Colp
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
Joshua Colp
4aba5a78d9
Merged revisions 50433 via svnmerge from
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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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2007-01-10 20:26:55 +00:00
Kevin P. Fleming
cd73a483f1
const-ify some more APIs, and fix rev 49710 from branch-1.4 in a better way here
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49711 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-05 23:32:42 +00:00
Steve Murphy
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-05 01:40:06 +00:00
Joshua Colp
5345469008
Documentation updates (thanks Shaun for the speechrec.txt one!)
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2006-08-24 15:44:24 +00:00
Kevin P. Fleming
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
Joshua Colp
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-18 16:22:26 +00:00
Russell Bryant
558d3924ac
don't leak a frame when breaking out of the loop on a timeout
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@33448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-11 14:52:04 +00:00
Kevin P. Fleming
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
Kevin P. Fleming
18606233da
fix various typos and other bits (from Ian Kinner)
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2006-05-30 16:01:50 +00:00
Joshua Colp
f35e669a98
Nothing to see here... move along
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30131 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-24 23:08:39 +00:00
Joshua Colp
6f8c704e2b
Update some documentation (file internal brain bug #42 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-24 22:00:37 +00:00
Joshua Colp
80a705c121
Make sure that the channel is answered before doing SpeechBackground. (issue #josh-wait-I-dont-have-issue-numbers)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27194 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-15 15:30:21 +00:00
Russell Bryant
04ecb29d03
remove almost all of the checks of the result from ast_strdupa() or alloca().
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As it turns out, all of these checks were useless, because alloca will never
return NULL.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-10 13:22:15 +00:00
Russell Bryant
48b812829d
remove \n from the end of a couple of synopsis fields
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@23929 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-01 14:57:48 +00:00
Luigi Rizzo
e43bc6634d
This rather large commit changes the way modules are loaded.
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As partly documented in loader.c and include/asterisk/module.h,
modules are now expected to return all of their methods and flags
into a structure 'mod_data', and are normally loaded with RTLD_NOW
| RTLD_LOCAL, so symbols are resolved immediately and conflicts
should be less likely. Only in a small number of cases (res_*,
typically) modules are loaded RTLD_GLOBAL, so they can export
symbols.
The core of the change is only the two files loader.c and
include/asterisk/module.h, all the rest is simply adaptation of the
existing modules to the new API, a rather mechanical (but believe
me, time and finger-consuming!) process whose detail you can figure
out by svn diff'ing any single module.
Expect some minor compilation issue after this change, please
report it on mantis http://bugs.digium.com/view.php?id=6968
so we collect all the feedback in one place.
I am just sorry that this change missed SVN version number 20000!
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2006-04-14 14:08:19 +00:00
Joshua Colp
5e43f18529
Updates to speech recognition API and dialplan utilities. Moved to using dialplan functions, and some other misc things.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@19645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-04-13 00:18:52 +00:00