Richard Mudgett
2758cc76e5
datastores: Audit ast_channel_datastore_remove usage.
...
Audit of v1.8 usage of ast_channel_datastore_remove() for datastore memory
leaks.
* Fixed leaks in app_speech_utils and func_frame_trace.
* Fixed app_speech_utils not locking the channel when accessing the
channel datastore list.
Review: https://reviewboard.asterisk.org/r/3859/
Audit of v11 usage of ast_channel_datastore_remove() for datastore memory
leaks.
* Fixed leak in func_jitterbuffer. (Was not in v12)
Review: https://reviewboard.asterisk.org/r/3860/
Audit of v12 usage of ast_channel_datastore_remove() for datastore memory
leaks.
* Fixed leaks in abstract_jb.
* Fixed leak in ast_channel_unsuppress(). Used by ARI mute control and
res_mutestream.
* Fixed ref leak in ast_channel_suppress(). Used by ARI mute control and
res_mutestream.
Review: https://reviewboard.asterisk.org/r/3861/
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Merged revisions 419684 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 419685 from http://svn.asterisk.org/svn/asterisk/branches/11
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2014-07-28 18:58:43 +00:00
Mark Michelson
dcf1ad14da
Add module support level to ast_module_info structure. Print it in CLI "module show" .
...
ASTERISK-23919 #close
Reported by Malcolm Davenport
Review: https://reviewboard.asterisk.org/r/3802
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-25 16:47:17 +00:00
Matthew Jordan
a2c912e997
media formats: re-architect handling of media for performance improvements
...
In the old times media formats were represented using a bit field. This was
fast but had a few limitations.
1. Asterisk was limited in how many formats it could handle.
2. Formats, being a bit field, could not include any attribute information.
A format was strictly its type, e.g., "this is ulaw".
This was changed in Asterisk 10 (see
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for
notes on that work) which led to the creation of the ast_format structure.
This structure allowed Asterisk to handle attributes and bundle information
with a format.
Additionally, ast_format_cap was created to act as a container for multiple
formats that, together, formed the capability of some entity. Another
mechanism was added to allow logic to be registered which performed format
attribute negotiation. Everywhere throughout the codebase Asterisk was
changed to use this strategy.
Unfortunately, in software, there is no free lunch. These new capabilities
came at a cost.
Performance analysis and profiling showed that we spend an inordinate
amount of time comparing, copying, and generally manipulating formats and
their related structures. Basic prototyping has shown that a reasonably
large performance improvement could be made in this area. This patch is the
result of that project, which overhauled the media format architecture
and its usage in Asterisk to improve performance.
Generally, the new philosophy for handling formats is as follows:
* The ast_format structure is reference counted. This removed a large amount
of the memory allocations and copying that was done in prior versions.
* In order to prevent race conditions while keeping things performant, the
ast_format structure is immutable by convention and lock-free. Violate this
tenet at your peril!
* Because formats are reference counted, codecs are also reference counted.
The Asterisk core generally provides built-in codecs and caches the
ast_format structures created to represent them. Generally, to prevent
inordinate amounts of module reference bumping, codecs and formats can be
added at run-time but cannot be removed.
* All compatibility with the bit field representation of codecs/formats has
been moved to a compatibility API. The primary user of this representation
is chan_iax2, which must continue to maintain its bit-field usage of formats
for interoperability concerns.
* When a format is negotiated with attributes, or when a format cannot be
represented by one of the cached formats, a new format object is created or
cloned from an existing format. That format may have the same codec
underlying it, but is a different format than a version of the format with
different attributes or without attributes.
* While formats are reference counted objects, the reference count maintained
on the format should be manipulated with care. Formats are generally cached
and will persist for the lifetime of Asterisk and do not explicitly need
to have their lifetime modified. An exception to this is when the user of a
format does not know where the format came from *and* the user may outlive
the provider of the format. This occurs, for example, when a format is read
from a channel: the channel may have a format with attributes (hence,
non-cached) and the user of the format may last longer than the channel (if
the reference to the channel is released prior to the format's reference).
For more information on this work, see the API design notes:
https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite
Finally, this work was the culmination of a large number of developer's
efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the
work in the Asterisk core, chan_sip, and was an invaluable resource in peer
reviews throughout this project.
There were a substantial number of patches contributed during this work; the
following issues/patch names simply reflect some of the work (and will cause
the release scripts to give attribution to the individuals who work on them).
Reviews:
https://reviewboard.asterisk.org/r/3814
https://reviewboard.asterisk.org/r/3808
https://reviewboard.asterisk.org/r/3805
https://reviewboard.asterisk.org/r/3803
https://reviewboard.asterisk.org/r/3801
https://reviewboard.asterisk.org/r/3798
https://reviewboard.asterisk.org/r/3800
https://reviewboard.asterisk.org/r/3794
https://reviewboard.asterisk.org/r/3793
https://reviewboard.asterisk.org/r/3792
https://reviewboard.asterisk.org/r/3791
https://reviewboard.asterisk.org/r/3790
https://reviewboard.asterisk.org/r/3789
https://reviewboard.asterisk.org/r/3788
https://reviewboard.asterisk.org/r/3787
https://reviewboard.asterisk.org/r/3786
https://reviewboard.asterisk.org/r/3784
https://reviewboard.asterisk.org/r/3783
https://reviewboard.asterisk.org/r/3778
https://reviewboard.asterisk.org/r/3774
https://reviewboard.asterisk.org/r/3775
https://reviewboard.asterisk.org/r/3772
https://reviewboard.asterisk.org/r/3761
https://reviewboard.asterisk.org/r/3754
https://reviewboard.asterisk.org/r/3753
https://reviewboard.asterisk.org/r/3751
https://reviewboard.asterisk.org/r/3750
https://reviewboard.asterisk.org/r/3748
https://reviewboard.asterisk.org/r/3747
https://reviewboard.asterisk.org/r/3746
https://reviewboard.asterisk.org/r/3742
https://reviewboard.asterisk.org/r/3740
https://reviewboard.asterisk.org/r/3739
https://reviewboard.asterisk.org/r/3738
https://reviewboard.asterisk.org/r/3737
https://reviewboard.asterisk.org/r/3736
https://reviewboard.asterisk.org/r/3734
https://reviewboard.asterisk.org/r/3722
https://reviewboard.asterisk.org/r/3713
https://reviewboard.asterisk.org/r/3703
https://reviewboard.asterisk.org/r/3689
https://reviewboard.asterisk.org/r/3687
https://reviewboard.asterisk.org/r/3674
https://reviewboard.asterisk.org/r/3671
https://reviewboard.asterisk.org/r/3667
https://reviewboard.asterisk.org/r/3665
https://reviewboard.asterisk.org/r/3625
https://reviewboard.asterisk.org/r/3602
https://reviewboard.asterisk.org/r/3519
https://reviewboard.asterisk.org/r/3518
https://reviewboard.asterisk.org/r/3516
https://reviewboard.asterisk.org/r/3515
https://reviewboard.asterisk.org/r/3512
https://reviewboard.asterisk.org/r/3506
https://reviewboard.asterisk.org/r/3413
https://reviewboard.asterisk.org/r/3410
https://reviewboard.asterisk.org/r/3387
https://reviewboard.asterisk.org/r/3388
https://reviewboard.asterisk.org/r/3389
https://reviewboard.asterisk.org/r/3390
https://reviewboard.asterisk.org/r/3321
https://reviewboard.asterisk.org/r/3320
https://reviewboard.asterisk.org/r/3319
https://reviewboard.asterisk.org/r/3318
https://reviewboard.asterisk.org/r/3266
https://reviewboard.asterisk.org/r/3265
https://reviewboard.asterisk.org/r/3234
https://reviewboard.asterisk.org/r/3178
ASTERISK-23114 #close
Reported by: mjordan
media_formats_translation_core.diff uploaded by kharwell (License 6464)
rb3506.diff uploaded by mjordan (License 6283)
media_format_app_file.diff uploaded by kharwell (License 6464)
misc-2.diff uploaded by file (License 5000)
chan_mild-3.diff uploaded by file (License 5000)
chan_obscure.diff uploaded by file (License 5000)
jingle.diff uploaded by file (License 5000)
funcs.diff uploaded by file (License 5000)
formats.diff uploaded by file (License 5000)
core.diff uploaded by file (License 5000)
bridges.diff uploaded by file (License 5000)
mf-codecs-2.diff uploaded by file (License 5000)
mf-app_fax.diff uploaded by file (License 5000)
mf-apps-3.diff uploaded by file (License 5000)
media-formats-3.diff uploaded by file (License 5000)
ASTERISK-23715
rb3713.patch uploaded by coreyfarrell (License 5909)
rb3689.patch uploaded by mjordan (License 6283)
ASTERISK-23957
rb3722.patch uploaded by mjordan (License 6283)
mf-attributes-3.diff uploaded by file (License 5000)
ASTERISK-23958
Tested by: jrose
rb3822.patch uploaded by coreyfarrell (License 5909)
rb3800.patch uploaded by jrose (License 6182)
chan_sip.diff uploaded by mjordan (License 6283)
rb3747.patch uploaded by jrose (License 6182)
ASTERISK-23959 #close
Tested by: sgriepentrog, mjordan, coreyfarrell
sip_cleanup.diff uploaded by opticron (License 6273)
chan_sip_caps.diff uploaded by mjordan (License 6283)
rb3751.patch uploaded by coreyfarrell (License 5909)
chan_sip-3.diff uploaded by file (License 5000)
ASTERISK-23960 #close
Tested by: opticron
direct_media.diff uploaded by opticron (License 6273)
pjsip-direct-media.diff uploaded by file (License 5000)
format_cap_remove.diff uploaded by opticron (License 6273)
media_format_fixes.diff uploaded by opticron (License 6273)
chan_pjsip-2.diff uploaded by file (License 5000)
ASTERISK-23966 #close
Tested by: rmudgett
rb3803.patch uploaded by rmudgetti (License 5621)
chan_dahdi.diff uploaded by file (License 5000)
ASTERISK-24064 #close
Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose
rb3814.patch uploaded by rmudgett (License 5621)
moh_cleanup.diff uploaded by opticron (License 6273)
bridge_leak.diff uploaded by opticron (License 6273)
translate.diff uploaded by file (License 5000)
rb3795.patch uploaded by rmudgett (License 5621)
tls_fix.diff uploaded by mjordan (License 6283)
fax-mf-fix-2.diff uploaded by file (License 5000)
rtp_transfer_stuff uploaded by mjordan (License 6283)
rb3787.patch uploaded by rmudgett (License 5621)
media-formats-explicit-translate-format-3.diff uploaded by file (License 5000)
format_cache_case_fix.diff uploaded by opticron (License 6273)
rb3774.patch uploaded by rmudgett (License 5621)
rb3775.patch uploaded by rmudgett (License 5621)
rtp_engine_fix.diff uploaded by opticron (License 6273)
rtp_crash_fix.diff uploaded by opticron (License 6273)
rb3753.patch uploaded by mjordan (License 6283)
rb3750.patch uploaded by mjordan (License 6283)
rb3748.patch uploaded by rmudgett (License 5621)
media_format_fixes.diff uploaded by opticron (License 6273)
rb3740.patch uploaded by mjordan (License 6283)
rb3739.patch uploaded by mjordan (License 6283)
rb3734.patch uploaded by mjordan (License 6283)
rb3689.patch uploaded by mjordan (License 6283)
rb3674.patch uploaded by coreyfarrell (License 5909)
rb3671.patch uploaded by coreyfarrell (License 5909)
rb3667.patch uploaded by coreyfarrell (License 5909)
rb3665.patch uploaded by mjordan (License 6283)
rb3625.patch uploaded by coreyfarrell (License 5909)
rb3602.patch uploaded by coreyfarrell (License 5909)
format_compatibility-2.diff uploaded by file (License 5000)
core.diff uploaded by file (License 5000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-20 22:06:33 +00:00
Corey Farrell
fbe0dfaf44
Fix dialplan function NULL channel safety issues
...
(closes issue ASTERISK-23391)
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3386/
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Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 411314 from http://svn.asterisk.org/svn/asterisk/branches/11
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2014-03-27 19:21:44 +00:00
Richard Mudgett
10d4ed93ff
app_speech_utils: Fix unresolved symbol ast_speech_get_setting().
...
Fixes regression introduced by -r374096.
* Made res_speech.export.in export ast_* symbols instead of specific
functions.
* Made app_speech_utils.c declare that it is dependent upon res_speech.
(issue ASTERISK-17136)
Reported by: Richard Kenner
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2013-09-16 18:00:32 +00:00
Joshua Colp
0fc114dc65
Add support for retrieving engine specific settings using the speech API and from dialplan.
...
(closes issue ASTERISK-17136)
Reported by: kenner
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-01 12:29:04 +00:00
Terry Wilson
34d670f786
Document Speech* apps hangup on failure and suggest TryExec
...
The Speech API apps return -1 on failure, which will hang up the channel. This
may not be desirable behavior for some, but it isn't something that can be
changed without breaking people's dialplans or writing an option to all of the
Speech apps that does what TryExec already does. This patch documents the
hangup behavior of the apps, and suggests TryExec as the solution.
(closes issue AST-813)
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Merged revisions 362816 from http://svn.asterisk.org/svn/asterisk/branches/10
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2012-04-20 14:50:42 +00:00
Terry Wilson
786f5898d1
Finalize ast_channel opaquification
...
Review: https://reviewboard.asterisk.org/r/1786/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-13 18:20:34 +00:00
Terry Wilson
ebaf59a656
Opaquification for ast_format structs in struct ast_channel
...
Review: https://reviewboard.asterisk.org/r/1770/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-24 00:32:20 +00:00
Terry Wilson
57f42bd74f
ast_channel opaquification of pointers and integral types
...
Review: https://reviewboard.asterisk.org/r/1753/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-20 23:43:27 +00:00
Terry Wilson
99cae5b750
Opaquify channel stringfields
...
Continue channel opaque-ification by wrapping all of the stringfields.
Eventually, we will restrict what can actually set these variables, but
the purpose for now is to hide the implementation and keep people from
adding code that directly accesses the channel structure. Semantic
changes will follow afterward.
Review: https://reviewboard.asterisk.org/r/1661/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-24 20:12:09 +00:00
Leif Madsen
a525edea59
Merged revisions 328247 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.10
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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
https://origsvn.digium.com/svn/asterisk/branches/1.8
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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
David Vossel
c26c190711
Asterisk media architecture conversion - no more format bitfields
...
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: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-03 16:22:10 +00:00
Tilghman Lesher
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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2010-09-02 05:27:53 +00:00
Matthew Nicholson
d38c6459f5
Merged revisions 264334 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19 20:02:57 +00:00
Tilghman Lesher
d8e0c58437
Expand codec bitfield from 32 bits to 64 bits.
...
Reviewboard: https://reviewboard.asterisk.org/r/416/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04 14:05:12 +00:00
Kevin P. Fleming
e6b2e9a750
Const-ify the world (or at least a good part of it)
...
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
Mark Michelson
932cef55fd
Swap reversed timevals.
...
This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-01 23:25:23 +00:00
Joshua Colp
2e2494591e
Merged revisions 177383 via svnmerge from
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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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2009-02-19 16:38:41 +00:00
Eliel C. Sardanons
206fe71680
Move Speech* applications and functions documentation to XML.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-06 21:18:51 +00:00
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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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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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
...
- 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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
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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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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.
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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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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
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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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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
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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