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20 authors
  1. Mar 16, 2018
    • Alexander Traud's avatar
      BuildSystem: Enable Advanced Linux Sound Architecture (ALSA) in NetBSD. · 00789174
      Alexander Traud authored 
      In the script ./configure, AST_EXT_LIB_CHECK checks for external libraries. Some
libraries do not specify all their dependencies and require additional shared
libraries. In AST_EXT_LIB_CHECK, this is the fifth parameter. However, if a
library is specified there, it must exist on the platform, because ./configure
tries to compile/link/execute a small app using those statements. For example,
the library libdl.so is Linux specific and does not exist on BSD-like platforms.

Furthermore, no supported platform/version was found, which still (ever?)
requires those additional libraries. Therefore, they were simply removed.

Finally, this change adds the error code ESTRPIPE to the channel driver
chan_alsa for those platforms which lack it, again for example NetBSD.

ASTERISK-27720

Change-Id: I3b21f2135f6cbfac7590ccdc2df753257f426e0b
      00789174
  3. Dec 22, 2017
  5. Apr 12, 2017
    • George Joseph's avatar
      modules: change module LOAD_FAILUREs to LOAD_DECLINES · 747beb1e
      George Joseph authored 
      In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE.  This prevents asterisk from exiting
if a module can't be loaded.  If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.

A new API was added to logger: ast_is_logger_initialized().  This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout.  If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.

Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25
      747beb1e
  7. Feb 13, 2017
    • Sean Bright's avatar
      cli: Fix various CLI documentation and completion issues · 3f943737
      Sean Bright authored 
      * app_minivm: Use built-in completion facilities to complete optional
arguments.

* app_voicemail: Use built-in completion facilities to complete
optional arguments.

* app_confbridge: Add missing colons after 'Usage' text.

* chan_alsa: Use built-in completion facilities to complete optional
arguments.

* chan_sip: Use built-in completion facilities to complete optional
arguments. Add completions for 'load' for 'sip show user', 'sip show
peer', and 'sip qualify peer.'

* chan_skinny: Correct and extend completions for 'skinny reset' and
'skinny show line.'

* func_odbc: Correct completions for 'odbc read' and 'odbc write'

* main/astmm: Use built-in completion facilities to complete arguments
for 'memory' commands.

* main/bridge: Correct completions for 'bridge kick.'

* main/ccss: Use built-in completion facilities to complete arguments
for 'cc cancel' command.

* main/cli: Add 'all' completion for 'channel request hangup.' Correct
completions for 'core set debug channel.' Correct completions for 'core
show calls.'

* main/pbx_app: Remove redundant completions for 'core show
applications.'

* main/pbx_hangup_handler: Remove unused completions for 'core show
hanguphandlers all.'

* res_sorcery_memory_cache: Add completion for 'reload' argument of
'sorcery memory cache stale' and properly implement.

Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca
      3f943737
  9. Oct 27, 2016
    • Corey Farrell's avatar
      Remove ASTERISK_REGISTER_FILE. · a6e5bae3
      Corey Farrell authored 
      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
      a6e5bae3
  11. Nov 09, 2015
  13. May 13, 2015
  15. 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
  17. Jul 25, 2014
  19. Jul 20, 2014
  21. May 09, 2014
  23. Mar 07, 2014
    • Scott Griepentrog's avatar
      uniqueid: channel linkedid, ami, ari object creation with id's · 80ef9a21
      Scott Griepentrog authored 
      Much needed was a way to assign id to objects on creation, and
much change was necessary to accomplish it.  Channel uniqueids
and linkedids are split into separate string and creation time
components without breaking linkedid propgation.  This allowed
the uniqueid to be specified by the user interface - and those
values are now carried through to channel creation, adding the
assignedids value to every function in the chain including the
channel drivers. For local channels, the second channel can be
specified or left to default to a ;2 suffix of first.  In ARI,
bridge, playback, and snoop objects can also be created with a
specified uniqueid.

Along the way, the args order to allocating channels was fixed
in chan_mgcp and chan_gtalk, and linkedid is no longer lost as
masquerade occurs.

(closes issue ASTERISK-23120)
Review: https://reviewboard.asterisk.org/r/3191/
........

Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      80ef9a21
  25. Dec 18, 2013
  27. Dec 05, 2013
  29. Dec 03, 2013
  31. Oct 03, 2013
  33. Oct 02, 2013
  35. Apr 14, 2013
  37. Oct 14, 2012
  39. Oct 01, 2012
  41. Jul 31, 2012
  43. 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
  45. Apr 17, 2012
  47. Feb 24, 2012
  49. Feb 20, 2012
  51. Feb 13, 2012
  53. Feb 01, 2012
  55. Jan 24, 2012
  57. 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
  59. Sep 09, 2011
    • Matthew Jordan's avatar
      Merged revisions 335078 via svnmerge from · 8b5ba33f
      Matthew Jordan authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines
  
  Merged revisions 335064 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines
    
    Updated SIP 484 handling; added Incomplete control frame
    
    When a SIP phone uses the dial application and receives a 484 Address 
    Incomplete response, if overlapped dialing is enabled for SIP, then
    the 484 Address Incomplete is forwarded back to the SIP phone and the
    HANGUPCAUSE channel variable is set to 28.  Previously, the Incomplete
    application dialplan logic was automatically triggered; now, explicit
    dialplan usage of the application is required.
    
    Additionally, this patch adds a new AST_CONTOL_FRAME type called
    AST_CONTROL_INCOMPLETE.  If a channel driver receives this control frame,
    it is an indication that the dialplan expects more digits back from the
    device.  If the device supports overlap dialing it should attempt to 
    notify the device that the dialplan is waiting for more digits; otherwise,
    it can handle the frame in a manner appropriate to the channel driver.
    
    (closes issue ASTERISK-17288)
    Reported by: Mikael Carlsson
    Tested by: Matthew Jordan
    
    Review: https://reviewboard.asterisk.org/r/1416/
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      8b5ba33f
  61. Jul 14, 2011
  63. May 26, 2011
  65. May 05, 2011
    • Russell Bryant's avatar
      Merged revisions 317478 via svnmerge from · 09389749
      Russell Bryant authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) | 12 lines
  
  Fix some consistency issues with jitterbuffer config.
  
  Store the defaults noted in the sample config files in the jitterbuffer config
  data structure.  This makes the CLI commands that output these settings show
  the right thing.  Also only show the settings that are relevant in the settings
  CLI commands, based on which jitterbuffer is selected and whether it's enabled.
  
  (closes issue #19083)
  Reported by: rgagnon
  Patches:
        issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      09389749
  67. Feb 03, 2011
  69. Jul 20, 2010
  71. Mar 02, 2010
    • David Vossel's avatar
      fixes adaptive jitterbuffer configuration · 862ebf4d
      David Vossel authored 
      When configuring the adaptive jitterbuffer, the target_extra
value not only could not be set from the configuration, but was
not even being set to its proper default.  This value is required
in order for the adaptive jitterbuffer to work correctly.  To resolve
this a config option has been added to expose this value to the conf
files, and a default value is provided when no config specific value
is present.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      862ebf4d
  73. Nov 12, 2009
  75. Nov 04, 2009
  77. Jul 28, 2009