Again, since res_jabber/res_xmpp have duplicate APIs, their documentation ref
links have to specify which reference they're referring to. The various
documentation parsers can interpret the module attribute however they want
in order to construct the appropriate links.
........

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  r379209 | mjordan | 2013-01-16 09:27:44 -0600 (Wed, 16 Jan 2013) | 8 lines
  
  Add module tags to documentation for res_jabber/res_xmpp
  
  Since res_jabber/res_xmpp provide the same APIs (app/func/manager/etc.),
  the XML documentation for each needs to call out which module is providing
  the documentation. The module attribute has been added to the various XML
  fragments for this purpose.
........
  r379210 | mjordan | 2013-01-16 09:30:20 -0600 (Wed, 16 Jan 2013) | 4 lines
  
  Update the dtd to actually *support* the module attribute in all elements
  
  Mea culpa.
........

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* Whitespace, doc-blocks, spelling, case, missing and incorrect tags.
* Add cleanup to Makefile for the Doxygen configuration update
* Start updating Doxygen configuration for cleaner output
* Enable inclusion of configuration files into documentation
* remove mantisworkflow...
* update documentation README
* Add markup to Tilghman's email and talk with him about updating his email, he knows...
* no code changes on this commit other than the mentioned Makefile change

(issue ASTERISK-20259)


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In r294740, the CODING-GUIDELINES was removed from the doc folder in favor
of the content on the Asterisk wiki.  Some folks still look in the doc folder
initially for coding guideline suggestions; as such, this patch adds a
CODING-GUIDELINES file back into the doc folder.  The content of the file
merely points to the correct page on the Asterisk wiki where the coding
guidelines currently live.

(closes issue ASTERISK-20279)
Reported by: Andrew Latham
Patches:
  CODING-GUIDELINES.diff uploaded by Andrew Latham (license 5985)
........

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A number of applications/AMI commands in Asterisk have specific behavioral
differences depending on the resource or channel technology those
applications are executed on.  For example, the MessageSend application/
command is technology agnostic, but how the channel drivers that support
that functionality behave is dependant on the protocols and channel
driver implementation.  Prior to this patch, those details were either
documented in the application/command documentation itself, or were left
undocumented.

This patch adds a new element to the documentation schema, <info/>.  An info
node is essentially a piece of technology specific reference information that
can be included by any top level XML documentation node.  For example, the
MessageSend application can now include XMPP/SIP specific information, where
that technology specific information can be defined in chan_motif/res_xmpp/
chan_sip.  Likewise, that information can also be included in the MessageSend
AMI command.

Review: https://reviewboard.asterisk.org/r/2049




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This patch adds some basic documentation for a number of modules.  This
includes core source files in Asterisk (those in main), as well as
chan_agent, chan_dahdi, chan_local, sig_analog, and sig_pri.  The DTD
has also been updated to allow referencing of AMI commands.

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This patch adds the core changes necessary to support AMI event documentation
in the source files of Asterisk, and adds documentation to those AMI events
defined in the core application modules.  Event documentation is built from
the source by two new python scripts, located in build_tools:
get_documentation.py and post_process_documentation.py.

The get_documentation.py script mirrors the actions of the existing AWK
get_documentation scripts, except that it will scan the entirety of a source
file for Asterisk documentation.  Upon encountering it, if the documentation
happens to be an AMI event, it will attempt to extract information about the
event directly from the manager event macro calls that raise the event.  The
post_process_documentation.py script combines manager event instances that
are the same event but documented in multiple source files.  It generates
the final core-[lang].xml file.

As this process can take longer to complete than a typical 'make all', it
is only performed if a new make target, 'full', is chosen.

Review: https://reviewboard.asterisk.org/r/1967/

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................
  r340109 | mnicholson | 2011-10-10 09:15:41 -0500 (Mon, 10 Oct 2011) | 18 lines
  
  Merged revisions 340108 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) | 11 lines
    
    Load the proper XML documentation when multiple modules document the same application.
    
    This patch adds an optional "module" attribute to the XML documentation spec
    that allows the documentation processor to match apps with identical names from
    different modules to their documentation. This patch also fixes a number of
    bugs with the documentation processor and should make it a little more
    efficient. Support for multiple languages has also been properly implemented.
    
    ASTERISK-18130
    Review: https://reviewboard.asterisk.org/r/1485/
  ........
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  r331143 | qwell | 2011-08-09 10:59:54 -0500 (Tue, 09 Aug 2011) | 9 lines
  
  Merged revisions 331142 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r331142 | qwell | 2011-08-09 10:58:16 -0500 (Tue, 09 Aug 2011) | 1 line
    
    Regenerate asterisk man page from sgml.
  ........
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  r331139 | qwell | 2011-08-09 10:50:07 -0500 (Tue, 09 Aug 2011) | 19 lines
  
  Merged revisions 306999 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r306999 | lathama | 2011-02-08 14:22:35 -0600 (Tue, 08 Feb 2011) | 12 lines
    
    Documentation Updates
    
    Note default polling setting in voicemail.conf
    Add missing config to asterisk.conf
    Update manpage
    
    (issue #16505)
    Reported by: tzafrir
    Patches: 
          asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46)
    Tested by: lathama, tzafrir
  ........
................


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  r331138 | qwell | 2011-08-09 10:47:20 -0500 (Tue, 09 Aug 2011) | 1 line
  
  Revert merge of r306999, due to merge conflict.
........


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Note default polling setting in voicemail.conf
Add missing config to asterisk.conf
Update manpage

(issue #16505)
Reported by: tzafrir
Patches:
     asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46)
Tested by: lathama, tzafrir



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Start updates to the man pages.

(issue #16505)
Reported by: tzafrir
Tested by: lathama


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........
  r294745 | russell | 2010-11-11 16:17:57 -0600 (Thu, 11 Nov 2010) | 6 lines
  
  Remove CCSS architecture PDF.
  
  It has been moved to:
  
  https://wiki.asterisk.org/wiki/display/AST/CCSS+Architecture
........


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........
  r294740 | russell | 2010-11-11 16:13:38 -0600 (Thu, 11 Nov 2010) | 11 lines
  
  Remove most of the contents of the doc dir in favor of the wiki content.
  
  This merge does the following things:
  
   * Removes most of the contents from the doc/ directory in favor
     of the wiki - http://wiki.asterisk.org/
  
   * Updates the build_tools/prep_tarball script to know how to export
     the contents of the wiki in both PDF and plain text formats so that
     the documentation is still included in Asterisk release tarballs.
........


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  r291725 | russell | 2010-10-14 07:08:43 -0500 (Thu, 14 Oct 2010) | 2 lines
  
  Fix a typo - s/seucre/secure/
........


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  r288082 | rmudgett | 2010-09-21 16:03:28 -0500 (Tue, 21 Sep 2010) | 1 line
  
  Add note in party manipulation chapter on interception macros.
........


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  r286647 | rmudgett | 2010-09-14 10:30:49 -0500 (Tue, 14 Sep 2010) | 1 line
  
  Corrected documented CONNECTED_LINE and REDIRECTING party manipulation macro names.
........


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  r285992 | diruggles | 2010-09-10 09:13:16 -0400 (Fri, 10 Sep 2010) | 1 line
  
  Added missing documentation for ExternalIVR feature added in January 2010
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  r284698 | rmudgett | 2010-09-02 11:34:32 -0500 (Thu, 02 Sep 2010) | 5 lines

  Added documentation for CONNECTEDLINE and REDIRECTING functions.

  (closes issue #17808)
  Reported by: jtodd

  Review: https://reviewboard.asterisk.org/r/875/
........


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  r282470 | lmadsen | 2010-08-16 13:01:00 -0500 (Mon, 16 Aug 2010) | 15 lines
  
  Merged revisions 282469 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010) | 7 lines
    
    Add information about creating sounds files using
    the sounds tools publically available so that others can create their
    own sounds prompts using the same tools we use to generate sounds releases.
    This allows people creating their own prompts to sound consistent with
    the prompts available from the open source project.
    
    SWP-595
  ........
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  r280740 | tilghman | 2010-08-03 13:42:24 -0500 (Tue, 03 Aug 2010) | 9 lines
  
  Merged revisions 280739 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03 Aug 2010) | 2 lines
    
    Document -B and -W flags and regenerate manpage from sgml
  ........
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sip.conf configuration for the channel and for devices.

The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary,
like SIP proxys and SBCs, decrement this counter and detects when it reaches zero,
at which point the SIP request is nicely killed in a SIP-friendly way.

Review: https://reviewboard.asterisk.org/r/778/

Thanks to dvossel for the review and good advice.



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Change the documented pgsql schema to use "timestamp" instead of "time",
as the latter is only a time without a date.

Added some missing columns for cel's pgsql schema, and corrected spelling
on some others. Updated cel's uniqueid size to be the same as the cdr.
Added id column to cel's pgsql schema and updated code to allow unknown
columns to get their default value instead of forcing 0 or empty string.

Added microseconds to the timestamp cel logs to pgsql.

Review: https://reviewboard.asterisk.org/r/734


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........
  r272562 | tilghman | 2010-06-25 15:17:37 -0500 (Fri, 25 Jun 2010) | 5 lines
  
  Make the structure of the table specified before match the queries and results.
  
  (closes issue #17557)
   Reported by: cmaj
........


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(closes issue #17511)
Reported by: klaus3000
Patches:
      channelvariables.tex-patch.txt uploaded by klaus3000 (license 65)
Tested by: pabelanger



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(closes issue #15757)
 Reported by: Marquis
 Patches: 
       distributed_devstate-XMPP.txt uploaded by lmadsen (license 10)
 Tested by: Marquis, lmadsen, marcelloceschia
 
Review: https://reviewboard.asterisk.org/r/351/


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  r270078 | pabelanger | 2010-06-12 14:54:20 -0400 (Sat, 12 Jun 2010) | 2 lines
  
  Fix typo in example
........


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Review: https://reviewboard.asterisk.org/r/688/



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After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.

Original patch by mikma, updated for trunk and revised by me.

(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel

Review: https://reviewboard.asterisk.org/r/191/


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Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
  (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages

Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup

AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.

SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
  snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
  'snom_aoc_enabled' sip.conf option.

IAX AOC Support
- Natively supports AOC pass-through through the use of the new
  AST_CONTROL_AOC frame type

DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
  pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
  example usage:
  ;requests AOC-S, AOC-D, and AOC-E on call setup
  exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))

Review:	https://reviewboard.asterisk.org/r/552/


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It is possible to connect to the manager interface before all Asterisk modules
are loaded. To ensure that an application does not send AMI actions that might
require a module that has not yet loaded, the application can listen for the
FullyBooted manager event. It will be sent upon connection if all modules have
been loaded, or as soon as loading is complete. The event:

   Event: FullyBooted
   Privilege: system,all
   Status: Fully Booted

Review: https://reviewboard.asterisk.org/r/639/


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........
  r260569 | lmadsen | 2010-05-03 09:57:39 -0500 (Mon, 03 May 2010) | 1 line
  
  Minor typo pointed out by pabelanger on IRC.
........


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This module implements an abstraction for retrieving and exporting
asterisk data.
Developed by:
	Brett Bryant <brettbryant@gmail.com>
	Eliel C. Sardanons (LU1ALY) <eliels@gmail.com>
For the Google Summer of code 2009 Project.
Documentation can be found in doxygen format and inside the
header include/asterisk/data.h

Review: https://reviewboard.asterisk.org/r/275/



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