Commit Graph

13 Commits

Author SHA1 Message Date
Naveen Albert bcc18ca9f5 general: Fix various typos.
ASTERISK-30089 #close

Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275
2022-07-12 07:46:03 -05:00
Naveen Albert 51d262af12 xmldocs: Improve examples.
Use example tags instead of regular para tags
where possible.

ASTERISK-30090

Change-Id: Iada8bbfda08f30b118cedf2d040bbb21e4966ec5
2022-06-09 03:47:41 -05:00
Corey Farrell a6e5bae3ef Remove ASTERISK_REGISTER_FILE.
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.

Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename

This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled.  This variable was only used in lock.c so it
is now initialized in that file only.

ASTERISK-26480 #close

Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
2016-10-27 09:53:55 -04:00
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
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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Merged revisions 411315 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-27 19:21:44 +00:00
Jason Parker c1a7567d24 ARI: Add support for suppressing media streams.
Also convert res_mutestream to use the core feature behind this.

(closes issue ASTERISK-21618)

Review: https://reviewboard.asterisk.org/r/2652/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-18 16:03:12 +00:00
David M. Lee c6cf12408e Fix the svn:keywords property on several files.
Normally I think keyword expansion is silly, but the one time it would have
been good, it didn't work because the property had quotes in it. This patch
fixes obviously busted svn:keywords properties.
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Merged revisions 385683 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 385689 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-15 16:10:10 +00:00
Richard Mudgett 3714e8b1e5 Convert MuteAudio documentation to XML.
* Added missing error exits with cause in manager_mutestream().

* Cleaned up manager_mutestream() and func_mute_write().

* Some whitespace and comment cleanup.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-19 20:26:51 +00:00
Terry Wilson 04da92c379 Replace direct access to channel name with accessor functions
There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.

This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.

The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.

The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).

Review: https://reviewboard.asterisk.org/r/1655/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
Leif Madsen a525edea59 Merged revisions 328247 via svnmerge from
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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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 d760e81f37 Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
   using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
   and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate.  This allows
   for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
   updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.

-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c

Review: https://reviewboard.asterisk.org/r/1104/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-02-22 23:04:49 +00:00
Julian Lyndon-Smith d85650e4aa Added MixMonitorMute manager command
Added a new manager command to mute/unmute MixMonitor audio on a channel. 
Added a new feature to audiohooks so that you can mute either read / write
(or both) types of frames - this allows for MixMonitor to mute either side
of the conversation without affecting the conversation itself.

(closes issue #16740)
Reported by: jmls

Review: https://reviewboard.asterisk.org/r/487/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-21 11:27:27 +00:00
Olle Johansson eca8f9082c Adding MUTEAUDIO() dialplan function and MuteAudio AMI action (pinepeach)
Review: https://reviewboard.asterisk.org/r/345/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-09-02 06:23:05 +00:00