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20 authors
  1. Apr 13, 2015
    • Matt Jordan's avatar
      git migration: Refactor the ASTERISK_FILE_VERSION macro · 4a582616
      Matt Jordan authored 
      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
      4a582616
  3. Mar 26, 2015
  5. Nov 06, 2014
  7. Sep 26, 2014
  9. Sep 16, 2014
  11. Aug 20, 2014
  13. Aug 14, 2014
  15. Aug 06, 2014
  17. Jul 20, 2014
  19. May 09, 2014
  21. Jan 27, 2014
  23. Oct 03, 2013
  25. Aug 29, 2013
  27. Aug 16, 2013
  29. Jul 08, 2013
  31. Jul 03, 2013
  33. Jun 24, 2013
    • Kinsey Moore's avatar
      Index installed sounds and implement ARI sounds queries · a0b7a49a
      Kinsey Moore authored 
      This adds support for stasis/sounds and stasis/sounds/{ID} queries via
the Asterisk RESTful Interface (ARI, formerly Stasis-HTTP).

The following changes have been made to accomplish this:
* A modular indexer was created for local media.
* A new function to get an ast_format associated with a file extension
  was added.  
* Modifications were made to the built-in HTTP server so that URI
  decoding could be deferred to the URI handler when necessary.
* The Stasis-HTTP sounds JSON documentation was modified to handle
  cases where multiple languages are installed in different formats.
* Register and Unregister events for formats were added to the system
  topic.

(closes issue ASTERISK-21584)
(closes issue ASTERISK-21585)
Review: https://reviewboard.asterisk.org/r/2507/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      a0b7a49a
  35. Jan 28, 2013
    • Russell Bryant's avatar
      Change cleanup ordering in filestream destructor. · 5d41d316
      Russell Bryant authored 
      This patch came about due to a problem observed where wav files had an
empty header.  The header is supposed to be updated in wav_close().  It
turns out that this was broken when the cache_record_files option from
asterisk.conf was enabled.  The cleanup code was moving the file to its
final destination *before* running the close() method of the file
destructor, so the header didn't get updated.

Another problem here is that the move was being done before actually
closing the FILE *.

Finally, the last bug fixed here is that I noticed that wav_close()
checks for stream->filename to be non-NULL.  In the previous cleanup
order, it's checking a pointer to freed memory.  This doesn't actually
cause anything to break, but it's treading on dangerous waters.  Now the
free() of stream->filename is happening after the format module's
close() method gets called, so it's safer.

Review: https://reviewboard.asterisk.org/r/2286/
........

Merged revisions 380210 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 380211 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      5d41d316
  37. Jan 22, 2013
    • Matthew Jordan's avatar
      Add ControlPlayback manager action · 7d9871b3
      Matthew Jordan authored 
      This patch adds the capability for asynchronous manipulation of audio being
played back to a channel though a new AMI action "ControlPlayback". The
ControlPlayback action supports a number of operations, the availability of
which depend on the application being used to send audio to the channel.
When the audio playback was initiated using the ControlPlayback application
or CONTROL STREAM FILE AGI command, the audio can be paused, stopped,
restarted, reversed, or skipped forward. When initiated by other mechanisms
(such as the Playback application), the audio can be stopped, reversed, or
skipped forward.

Review: https://reviewboard.asterisk.org/r/2265/

(closes issue ASTERISK-20882)
Reported by: mjordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7d9871b3
  39. Dec 11, 2012
  41. Aug 30, 2012
    • Matthew Jordan's avatar
      Clean up doxygen warnings · 8018b879
      Matthew Jordan authored 
      This patch fixes numerous doxygen warnings across Asterisk.  It also updates
the makefile to regenerate the doxygen configuration on the local system
before running doxygen to help prevent warnings/errors on the local system.

Much thanks to Andrew for tackling one of the Asterisk janitor projects!

(issue ASTERISK-20259)
Reported by: Andrew Latham
Patches:
  doxygen_partial.diff uploaded by Andrew Latham (license 5985)
  make_progdocs.diff uploaded by Andrew Latham (license 5985)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      8018b879
  43. Aug 21, 2012
  45. Jul 31, 2012
  47. Jun 15, 2012
    • Kevin P. Fleming's avatar
      Multiple revisions 369001-369002 · 166b4e2b
      Kevin P. Fleming authored 
      ........
  r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines
  
  Add support-level indications to many more source files.
  
  Since we now have tools that scan through the source tree looking for files
  with specific support levels, we need to ensure that every file that is
  a component of a 'core' or 'extended' module (or the main Asterisk binary)
  is explicitly marked with its support level. This patch adds support-level
  indications to many more source files in tree, but avoids adding them to
  third-party libraries that are included in the tree and to source files
  that don't end up involved in Asterisk itself.
........
  r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines
  
  Add a script to enable finding source files without support-levels defined.
........

Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 369005 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      166b4e2b
  49. Jun 04, 2012
    • Mark Michelson's avatar
      Merge changes dealing with support for Digium phones. · 14a98556
      Mark Michelson authored 
      Presence support has been added. This is accomplished by
allowing for presence hints in addition to device state
hints. A dialplan function called PRESENCE_STATE has been
added to allow for setting and reading presence. Presence
can be transmitted to Digium phones using custom XML
elements in a PIDF presence document.

Voicemail has new APIs that allow for moving, removing,
forwarding, and playing messages. Messages have had a new
unique message ID added to them so that the APIs will work
reliably. The state of a voicemail mailbox can be obtained
using an API that allows one to get a snapshot of the mailbox.
A voicemail Dialplan App called VoiceMailPlayMsg has been
added to be able to play back a specific message.

Configuration hooks have been added. Configuration hooks
allow for a piece of code to be executed when a specific
configuration file is loaded by a specific module. This is
useful for modules that are dependent on the configuration
of other modules.

chan_sip now has a public method that allows for a custom
SIP INFO request to be sent mid-dialog. Digium phones use
this in order to display progress bars when files are played.

Messaging support has been expanded a bit. The main
visible difference is the addition of an AMI action
MessageSend.

Finally, a ParkingLots manager action has been added in order
to get a list of parking lots.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      14a98556
  51. May 14, 2012
    • Kinsey Moore's avatar
      Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) · b5a6de76
      Kinsey Moore authored 
      This is the starting point for the Asterisk 11: Who Hung Up work and provides
a framework which will allow channel drivers to report the types of hangup
cause information available in SIP_CAUSE without incurring the overhead of the
MASTER_CHANNEL dialplan function. The initial implementation only includes
cause generation for chan_sip and does not include cause code translation
utilities.

This change deprecates SIP_CAUSE and replaces its method of reporting cause
codes with the new framework. This change also deprecates the 'storesipcause'
option in sip.conf.

Review: https://reviewboard.asterisk.org/r/1822/
(Closes issue SWP-4221)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      b5a6de76
  53. Mar 22, 2012
  55. Mar 13, 2012
  57. Mar 01, 2012
  59. Feb 29, 2012
  61. Feb 24, 2012
  63. Feb 20, 2012
  65. Feb 13, 2012
  67. Jan 24, 2012
  69. Jan 09, 2012
    • Terry Wilson's avatar
      Replace direct access to channel name with accessor functions · 04da92c3
      Terry Wilson authored 
      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
      04da92c3
  71. Jan 05, 2012
  73. Dec 22, 2011
  75. Nov 11, 2011
  77. Oct 03, 2011