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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 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
  5. Dec 09, 2013
  7. Feb 05, 2013
  9. May 18, 2012
    • Matthew Jordan's avatar
      Fix a variety of memory leaks · 7b513206
      Matthew Jordan authored 
      This patch addresses a number of memory leaks in a variety of modules that were
found by a static analysis tool.  A brief summary of the changes:

* app_minivm:       free ast_str objects on off nominal paths
* app_page:         free the ast_dial object if the requested channel technology
                    cannot be appended to the dialing structure
* app_queue:        if a penalty rule failed to match any existing rule list
                    names, the created rule would not be inserted and its memory
                    would be leaked
* app_read:         dispose of the created silence detector in the presence of
                    off nominal circumstances
* app_voicemail:    dispose of an allocated unique ID field for MWI event
                    un-subscribe requests in off nominal paths; dispose of
                    configuration objects when using the secret.conf option
* chan_dahdi:       dispose of the allocated frame produced by ast_dsp_process
* chan_iax2:        properly unref peer in CLI command "iax2 unregister"
* chan_sip:         dispose of the allocated frame produced by sip_rtp_read's
                    call of ast_dsp_process; free memory in parse unit tests
* func_dialgroup:   properly deref ao2 object grhead in nominal path of
                    dialgroup_read
* func_odbc:        free resultset in off nominal paths of odbc_read
* cli:              free match_list in off nominal paths of CLI match completion
* config:           free comment_buffer/list_buffer when configuration file load
                    is unchanged; free the same buffers any time they were
                    created and config files were processed
* data:             free XML nodes in various places
* enum:             free context buffer in off nominal paths
* features:         free ast_call_feature in off nominal paths of applicationmap
                    config processing
* netsock2:         users of ast_sockaddr_resolve pass in an ast_sockaddr struct
                    that is allocated by the method.  Failures in
                    ast_sockaddr_resolve could result in the users of the method
                    not knowing whether or not the buffer was allocated.  The
                    method will now not allocate the ast_sockaddr struct if it
                    will return failure.
* pbx:              cleanup hash table traversals in off nominal paths; free
                    ignore pattern buffer if it already exists for the specified
                    context
* xmldoc:           cleanup various nodes when we no longer need them
* main/editline:    various cleanup of pointers not being freed before being
                    assigned to other memory, cleanup along off nominal paths
* menuselect/mxml:  cleanup of value buffer for an attribute when that attribute
                    did not specify a value
* res_calendar*:    responses are allocated via the various *_request method
                    returns and should not be allocated in the various
                    write_event methods; ensure attendee buffer is freed if no
                    data exists in the parsed node; ensure that calendar objects
                    are de-ref'd appropriately
* res_jabber:       free buffer in off nominal path
* res_musiconhold:  close the DIR* object in off nominal paths
* res_rtp_asterisk: if we run out of ports, close the rtp socket object and free
                    the rtp object
* res_srtp:         if we fail to create the session in libsrtp, destroy the
                    temporary ast_srtp object

(issue ASTERISK-19665)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1922
........

Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 366881 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7b513206
  11. Mar 24, 2012
  13. Mar 10, 2012
  15. Feb 28, 2012
  17. Jan 24, 2012
  19. 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
  21. Jul 14, 2011
  23. Dec 10, 2009
    • Jeff Peeler's avatar
      Add audio announcement option to app_page · 2414bc80
      Jeff Peeler authored 
      As described in the CHANGES file:
* MeetMe has a new option 'G' to play an announcement before joining a
  conference.
* Page has a new option 'A(x)' which will playback an announcement 
  simultaneously to all paged phones (and optionally excluding the caller's one 
  using the new option 'n') before the call is bridged.

To add the new option to meetme, the conference flag options had to be extended 
to 64 bits.

(closes issue #14365)
Reported by: dferrer
Patches:
      page_announce.patch uploaded by dferrer (license 525)
      modified by me

Review: https://reviewboard.asterisk.org/r/188/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      2414bc80
  25. Jun 15, 2009
  27. May 21, 2009
    • Kevin P. Fleming's avatar
      Const-ify the world (or at least a good part of it) · e6b2e9a7
      Kevin P. Fleming authored 
      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
      e6b2e9a7
  29. May 12, 2009
  31. Jan 25, 2009
  33. Jan 14, 2009
    • Steve Murphy's avatar
      Merged revisions 168608 via svnmerge from · a4743411
      Steve Murphy authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1 line
  
  app_page was failing to compile in dev-mode on my gcc-4.2.4 system. This change gets rid of the warning.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      a4743411
    • Terry Wilson's avatar
      Merged revisions 168593 via svnmerge from · 2015f9c9
      Terry Wilson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) | 20 lines
  
  Don't overflow when paging more than 128 extensions
  
  The number of available slots for calls in app_page was hardcoded to 128.
  Proper bounds checking was not in place to enforce this limit, so if more than
  128 extensions were passed to the Page() app, Asterisk would crash.  This patch
  instead dynamically allocates memory for the ast_dial structures and removes
  the (non-functional) arbitrary limit.
  
  This issue would have special importance to anyone who is dynamically creating
  the argument passed to the Page application and allowing more than 128
  extensions to be added by an outside user via some external interface.
  
  The patch posted by a_villacis was slightly modified for some coding guidelines
  and other cleanups.  Thanks, a_villacis!
  (closes issue #14217)
  Reported by: a_villacis
  Patches: 
        20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660)
  Tested by: otherwiseguy
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      2015f9c9
  35. Dec 15, 2008
  37. Nov 05, 2008
  39. Nov 04, 2008
  41. Nov 02, 2008
  43. Nov 01, 2008
  45. Oct 31, 2008
  47. Jun 12, 2008
  49. Mar 11, 2008
  51. Nov 21, 2007
  53. Nov 19, 2007
  55. Nov 16, 2007
    • Luigi Rizzo's avatar
      Start untangling header inclusion in a way that does not affect · fdb7f7ba
      Luigi Rizzo authored 
      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
      fdb7f7ba
  57. Jul 31, 2007
  59. Jul 23, 2007
  61. Jul 16, 2007
  63. Jun 01, 2007
  65. Mar 16, 2007
  67. Feb 12, 2007
  69. Jan 24, 2007
  71. Jan 18, 2007