Rodrigo Ramírez Norambuena 
							
						 
					 
					
						
						
							
						
						eec010829a 
					 
					
						
						
							
							AST_MODULE_INFO: Format corrections to the usages of AST_MODULE_INFO macro.  
						
						... 
						
						
						
						Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723 
						
						
					 
					
						2015-05-13 16:34:23 -05:00 
						 
				 
			
				
					
						
							
							
								Matt Jordan 
							
						 
					 
					
						
						
							
						
						4a58261694 
					 
					
						
						
							
							git migration: Refactor the ASTERISK_FILE_VERSION macro  
						
						... 
						
						
						
						Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.
Specifically, it does the following:
* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
  remove passing the version in with the macro. Other facilities
  than 'core show file version' make use of the file names, such as
  setting a debug level only on a specific file. As such, the act of
  registering source files with the Asterisk core still has use. The
  macro rename now reflects the new macro purpose.
* main/asterisk:
  - Refactor the file_version structure to reflect that it no longer
    tracks a version field.
  - Remove the "core show file version" CLI command. Without the file
    version, it is no longer useful.
  - Remove the ast_file_version_find function. The file version is no
    longer tracked.
  - Rename ast_register_file_version/ast_unregister_file_version to
    ast_register_file/ast_unregister_file, respectively.
* main/manager: Remove value from the Version key of the ModuleCheck
  Action. The actual key itself has not been removed, as doing so would
  absolutely constitute a backwards incompatible change. However, since
  the file version is no longer tracked, there is no need to attempt to
  include it in the Version key.
* UPGRADE: Add notes for:
  - Modification to the ModuleCheck AMI Action
  - Removal of the "core show file version" CLI command
Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e 
						
						
					 
					
						2015-04-13 03:48:57 -04:00 
						 
				 
			
				
					
						
							
							
								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/ 
........
Merged revisions 419684 from http://svn.asterisk.org/svn/asterisk/branches/1.8 
........
Merged revisions 419685 from http://svn.asterisk.org/svn/asterisk/branches/11 
........
Merged revisions 419686 from http://svn.asterisk.org/svn/asterisk/branches/12 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419688  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						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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........
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........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						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
........
Merged revisions 399197 from http://svn.asterisk.org/svn/asterisk/branches/12 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399198  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362817  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						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  
						
						... 
						
						
						
						https://origsvn.digium.com/svn/asterisk/branches/1.10 
................
  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 
  
  ........
    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 
  ........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
					
						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  
						
						... 
						
						
						
						https://origsvn.digium.com/svn/asterisk/branches/1.8 
........
  r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines
  
  When optional_api is non-optional, force dependent modules to be loaded.
  
  (closes issue #17707 )
   Reported by: ira
   Patches: 
         20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14)
   Tested by: tilghman
   
  Review: https://reviewboard.asterisk.org/r/876/ 
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284628  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
					
						2010-09-02 05:27:53 +00:00 
						 
				 
			
				
					
						
							
							
								Matthew Nicholson 
							
						 
					 
					
						
						
							
						
						d38c6459f5 
					 
					
						
						
							
							Merged revisions 264334 via svnmerge from  
						
						... 
						
						
						
						https://origsvn.digium.com/svn/asterisk/branches/1.4 
........
  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
........
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.  
						
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						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)  
						
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						This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/ 
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
						
						
					 
					
						2009-05-21 21:13:09 +00:00 
						 
				 
			
				
					
						
							
							
								Mark Michelson 
							
						 
					 
					
						
						
							
						
						932cef55fd 
					 
					
						
						
							
							Swap reversed timevals.  
						
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						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  
						
						... 
						
						
						
						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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177384  65c4cc65-6c06-0410-ace0-fbb531ad65f3 
					
						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  
						
						... 
						
						
						
						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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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  
						
						... 
						
						
						
						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  
						
						... 
						
						
						
						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