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44 Commits

Author SHA1 Message Date
Russell Bryant
22fae48e3c Add the 'n' option to SpeechBackground, which has the application not answer the 
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 
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 
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. 
(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 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-08-13 21:59:15 +00:00
Joshua Colp
3dac92ecbf Merged revisions 79207 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 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. 
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 
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. 
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. 
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. 
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. 
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 
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
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 
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() 
(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 
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 
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-04-13 18:09:29 +00:00
Joshua Colp
4b618442a0 Merged revisions 60361 via svnmerge from 
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-04-06 01:15:50 +00:00
Joshua Colp
d6f029b4f7 Merged revisions 59963 via svnmerge from 
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 
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 
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 
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 
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 
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 
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-02-08 17:56:53 +00:00
Joshua Colp
68d39c2b72 Merged revisions 51251 via svnmerge from 
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2007-01-18 19:19:24 +00:00
Joshua Colp
4aba5a78d9 Merged revisions 50433 via svnmerge from 
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 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 
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. 
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!) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-08-24 15:44:24 +00:00
Kevin P. Fleming
0a27d8bfe5 merge new_loader_completion branch, including (at least): 
- 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: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 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 
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 
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 :-) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-06-07 18:54:56 +00:00
Kevin P. Fleming
18606233da fix various typos and other bits (from Ian Kinner) 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-05-30 16:01:50 +00:00
Joshua Colp
f35e669a98 Nothing to see here... move along 
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) 
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) 
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(). 
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 
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. 
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!



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 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. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@19645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-13 00:18:52 +00:00
Joshua Colp
416cd5d8a0 Add ability to see if the person calling said anything or not. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@19512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-12 19:57:21 +00:00
Joshua Colp
e4811a8248 Presenting a revised data stores and oh my, a generic speech recognition API! I wonder what we can do with this now... 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 2006-04-10 23:29:50 +00:00