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20 authors
  1. Aug 02, 2013
  3. Feb 19, 2013
  5. 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
  7. Apr 19, 2012
  9. Mar 13, 2012
  11. Mar 01, 2012
  13. Feb 29, 2012
  15. Feb 20, 2012
  17. Feb 13, 2012
  19. Jan 27, 2012
  21. Jan 24, 2012
  23. 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
  25. Jul 14, 2010
    • Richard Mudgett's avatar
      Expand the caller ANI field to an ast_party_id · cf7bbcc4
      Richard Mudgett authored 
      Expand the ani field in ast_party_caller and ast_party_connected_line to
an ast_party_id.

This is an extension to the ast_callerid restructuring patch in review:
https://reviewboard.asterisk.org/r/702/

Review: https://reviewboard.asterisk.org/r/744/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      cf7bbcc4
    • Richard Mudgett's avatar
      Make compile again. · 2cf60bb0
      Richard Mudgett authored 
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      2cf60bb0
    • Richard Mudgett's avatar
      ast_callerid restructuring · ec37ffbd
      Richard Mudgett authored 
      The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.

Eliminate struct ast_callerid and replace it with the following struct
organization:

struct ast_party_name {
	char *str;
	int char_set;
	int presentation;
	unsigned char valid;
};
struct ast_party_number {
	char *str;
	int plan;
	int presentation;
	unsigned char valid;
};
struct ast_party_subaddress {
	char *str;
	int type;
	unsigned char odd_even_indicator;
	unsigned char valid;
};
struct ast_party_id {
	struct ast_party_name name;
	struct ast_party_number number;
	struct ast_party_subaddress subaddress;
	char *tag;
};
struct ast_party_dialed {
	struct {
		char *str;
		int plan;
	} number;
	struct ast_party_subaddress subaddress;
	int transit_network_select;
};
struct ast_party_caller {
	struct ast_party_id id;
	char *ani;
	int ani2;
};

The new organization adds some new information as well.

* The party name and number now have their own presentation value that can
be manipulated independently.  ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.

* The party name and number now have a valid flag.  Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.

* The party name now has a character set value.  SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.

* The dialed party now has a numbering plan value that could be useful to
have available.

The various channel drivers will need to be updated to support the new
core features as needed.  They have simply been converted to supply
current functionality at this time.


The following items of note were either corrected or enhanced:

* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.

* CALLERPRES() is now deprecated because the name and number have their
own presentation values.

* Fixed app_alarmreceiver.c write_metadata().  The workstring[] could
contain garbage.  It also can only contain the caller id number so using
ast_callerid_parse() on it is silly.  There was also a typo in the
CALLERNAME if test.

* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string.  ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string.  Then using
ast_shrink_phone_number() could alter it even more.

* Fixed caller ID name and number memory leak in chan_usbradio.c.

* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.

* Protected access to a caller channel with lock in chan_sip.c.

* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk().  Also made save all caller ID data instead of just the name
and number strings.

* Simplified cdr.c set_one_cid().  It hand coded the ast_callerid_merge()
function.

* Corrected some weirdness with app_privacy.c's use of caller
presentation.

Review:	https://reviewboard.asterisk.org/r/702/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      ec37ffbd
  27. Apr 03, 2010
  29. Oct 21, 2009
    • Kevin P. Fleming's avatar
      Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it. · cdd1f9e2
      Kevin P. Fleming authored 
      This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the
case where multiple results need to be returned; OBJ_NODATA mode
already was supported). In addition, it converts ast_channel_iterators
(only the targeted versions, not the ones that iterate over all
channels) to use this method.

During this work, I removed the 'ao2_flags' arguments to the
ast_channel_iterator constructor functions; there were no uses of that
argument yet, there is only one possible flag to pass, and it made the
iterators less 'opaque'. If at some point in the future someone really
needs an ast_channel_iterator that does not lock the container, we can
provide constructor(s) for that purpose.

Review: https://reviewboard.asterisk.org/r/379/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      cdd1f9e2
  31. Aug 10, 2009
  33. Apr 24, 2009
    • Russell Bryant's avatar
      Convert the ast_channel data structure over to the astobj2 framework. · cba19c8a
      Russell Bryant authored 
      There is a lot that could be said about this, but the patch is a big 
improvement for performance, stability, code maintainability, 
and ease of future code development.

The channel list is no longer an unsorted linked list.  The main container 
for channels is an astobj2 hash table.  All of the code related to searching 
for channels or iterating active channels has been rewritten.  Let n be 
the number of active channels.  Iterating the channel list has gone from 
O(n^2) to O(n).  Searching for a channel by name went from O(n) to O(1).  
Searching for a channel by extension is still O(n), but uses a new method 
for doing so, which is more efficient.

The ast_channel object is now a reference counted object.  The benefits 
here are plentiful.  Some benefits directly related to issues in the 
previous code include:

1) When threads other than the channel thread owning a channel wanted 
   access to a channel, it had to hold the lock on it to ensure that it didn't 
   go away.  This is no longer a requirement.  Holding a reference is 
   sufficient.

2) There are places that now require less dealing with channel locks.

3) There are places where channel locks are held for much shorter periods 
   of time.

4) There are places where dealing with more than one channel at a time becomes 
   _MUCH_ easier.  ChanSpy is a great example of this.  Writing code in the 
   future that deals with multiple channels will be much easier.

Some additional information regarding channel locking and reference count 
handling can be found in channel.h, where a new section has been added that 
discusses some of the rules associated with it.

Mark Michelson also assisted with the development of this patch.  He did the 
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it 
much easier to deal with holding on to a channel pointer for an extended period 
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.

Thanks to David Vossel for his assistance with this branch, as well.  David 
did the conversion of the DAHDIScan application by making it become a wrapper 
for ChanSpy internally.

The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.

Review: http://reviewboard.digium.com/r/203/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      cba19c8a
  35. Feb 17, 2009
    • Russell Bryant's avatar
      Merge a large set of updates to the Asterisk indications API. · 4ec30136
      Russell Bryant authored 
      This patch includes a number of changes to the indications API.  The primary
motivation for this work was to improve stability.  The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.

The changes included are:

1) Remove the module res_indications.  This included the critical functionality
   that actually loaded the indications configuration.  I have seen many people
   have Asterisk problems because they accidentally did not have an
   indications.conf present and loaded.  Now, this code is in the core,
   and Asterisk will fail to start without indications configuration.

   There was one part of res_indications, the dialplan applications, which did
   belong in a module, and have been moved to a new module, app_playtones.

2) Object management has been significantly changed.  Tone zones are now
   managed using astobj2, and it is no longer possible to crash Asterisk by
   issuing a reload that destroys tone zones while they are in use.

3) The API documentation has been filled out.

4) The API has been updated to follow our naming conventions.

5) Various bits of code throughout the tree have been updated to account
   for the API update.

6) Configuration parsing has been mostly re-written.

7) "Code cleanup"

The code is from svn/asterisk/team/russell/indications/.

Review: http://reviewboard.digium.com/r/149/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4ec30136
  37. Feb 07, 2009
  39. Jan 13, 2009
  41. Dec 13, 2008
  43. Dec 10, 2008
  45. May 02, 2008
  47. Mar 03, 2008
  49. Feb 26, 2008
  51. Jan 05, 2008
  53. Dec 20, 2007
  55. Dec 19, 2007
    • Luigi Rizzo's avatar
      make netsmp build under AST_DEVMODE. Description, included in the source, · 094ec652
      Luigi Rizzo authored 
      is below. I should note that the PACKAGE_* macros that asterisk
defines in autoconfig.h are not used anywhere in the tree so
they should just be removed.

 /*
  * There is some collision collision between netsmp and asterisk names,
  * causing build under AST_DEVMODE to fail.
  *
  * The following PACKAGE_* macros are one place.
  * Also netsnmp has an improper check for HAVE_DMALLOC_H, using
  *    #if HAVE_DMALLOC_H   instead of #ifdef HAVE_DMALLOC_H
  * As a countermeasure we define it to 0, however this will fail
  * when the proper check is implemented.
  */

No


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93875 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      094ec652
  57. Dec 14, 2007
  59. Dec 07, 2007
  61. Aug 23, 2007
  63. Jul 18, 2007
  65. Jun 13, 2007
  67. Jun 08, 2007
  69. Dec 25, 2006
  71. Dec 19, 2006