* Consistently use spaces in rest-api-templates/asterisk_processor.py.
* Exclude third-party from docs/full-en_US.xml.
* Add docs/full-en_US.xml to .gitignore.
* Use list() to convert python3 view.
* Use python3 print function.
* Replace cmp() with equivalent equation.
* Replace reference to out of scope subtype variable with name
  parameter.
* Use unescaping triple bracket notation in mustache templates where
  needed.  This causes behavior of Python2 to be maintained when using
  Python3.
* Fix references to has_websocket / is_websocket in
  res_ari_resource.c.mustache.
* Update calculation of has_websocket to use any().
* Use unicode mode for writing output file in transform.py.
* Replace 'from swagger_model import *' with explicit import of required
  symbols.

I have not tested spandspflow2pcap.py or voicemailpwcheck.py, only the
print syntax has been fixed.

Change-Id: If5c5b556a2800d41a3e2cfef080ac2e151178c33

This document is out of date and is superseded by content on the
Asterisk wiki.

ASTERISK-24386 #close

Change-Id: Idbf95b27b096c205251e1bbb560c79224ba81822

Change-Id: I8f494b0c895a69b8bc94656d0c6ceebecb0394d8

Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0

Change-Id: I1bc7957121cc7ae27dca04acc3613f4e1858476a

We should be sending people to secure web URL's where available.
Update README's and docs.

Change-Id: Id5b1e049b0b18b49a784f1254605aefa244ce19a

In multi-party bridges, Asterisk currently supports two video modes:
 * Follow the talker, in which the speaker with the most energy is shown
   to all participants but the speaker, and the speaker sees the
   previous video source
 * Explicitly set video sources, in which all participants see a locked
   video source

Prior to this patch, ARI had no ability to manipulate the video source.
This isn't important for two-party bridges, in which Asterisk merely
relays the video between the participants. However, in a multi-party
bridge, it can be advantageous to allow an external application to
manipulate the video source.

This patch provides two new routes to accomplish this:
(1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId}
    Sets a video source to an explicit channel
(2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource
    Removes any explicit video source, and sets the video mode to talk
    detection

ASTERISK-26595 #close

Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621

This updates func_channel.c and main/message.c to use a generic xpointer
include instead of including info from each channel driver.  Now the
name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in
documentation for func_channel.  Setting the name attribute of info to
MessageToInfo or MessageFromInfo causes it to be included in the
MessageSend application and AMI action.

Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea

* Following the example of the PJSIP channel driver, the channel
  technology specific documentation has been moved to the respective
  channel drivers that provide that functionality. This has the benefit
  of locating the documentation of items with those modules that provide
  it.

* Examples of using the CHANNEL function for both standard items as well
  as for PJSIP have been added.

* The 'max_forwards' standard item has been documented.

Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b

We don't need pjproject's documentation embedded in Asterisk's.

Change-Id: Iea6f5a621c0f4e3168dda3321eaab258d9f24a17

ASTERISK-25189 #close
Reported by: John Hardin

Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac

Previous versions of Asterisk processed command-line options before
processing asterisk.conf.  This meant that if an option was set in
asterisk.conf, it could not be overridden with the equivelent command
line option.  This change causes Asterisk to process the command-line
twice.  First it processes options that are needed to load asterisk.conf,
then it processes the remaining options after the config is read.

This changes the function of -X slightly.  Previously using -X without
disabling execincludes in asterisk.conf caused #exec to be usable in any
config.  Now -X only enables #exec for the load of asterisk.conf, if it
is wanted in the rest of the system it must be enabled with execincludes
in asterisk.conf.  Updated 'asterisk -h' and 'man asterisk' to reflect
the limited function of -X.

ASTERISK-25042 #close
Reported by: Corey Farrell

Change-Id: I1450d45c15b4467274b871914d893ed4f6564cd7

Moved contrib/asterisk-ng-doxygen to doc/asterisk-ng-doxygen.in

Changed /Makefile to copy asterisk-ng-doxygen.in to
asterisk-ng-doxygen then modify it with version instead of
modifying asterisk-ng-doxygen directly.  Updated clean
targets as well.

Updated /.gitignore and doc/.gitignore.

Change-Id: I38712d3e334fa4baec19d30d05de8c6f28137622

Create the directory './doc/rest-api' at the start of 'make ari-stubs'
to prevent an error when documentation is generated.  The directory is
also added to git ignores.

ASTERISK-25027
Reported by: Corey Farrell

Change-Id: Iaccc7f0138501c23aa78feaca2f3cce9e68cbc1b

Add the .gitignore and .gitreview files to the asterisk repo.

NB:  You can add local ignores to the .git/info/exclude file
without having to do a commit.

Common ignore patterns are in the top-level .gitignore file.
Subdirectory-specific ignore patterns are in their own .gitignore
files.

Change-Id: I842a1588ff27d8a0189f12d597f0a7af033d6c69
Tested-by: George Joseph

ASTERISK-23768 #close
Reported by: Jeremy Lainé
Patches:
  escape_manpage_hyphen.patch uploaded by Jeremy Lainé (License #6561)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423919 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a manpage for the smsq utility. Note that this is one of
the patches the Debian distro applies for the Asterisk project, as per
ASTERISK-24191.

Review: https://reviewboard.asterisk.org/r/3895/

ASTERISK-24171 #close
Reported by: Jeremy Laine
patches:
  smsq.8 uploaded by Jeremy Laine (License 6561)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422380 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a manpage for the aelparse utility. Note that this is one of
the patches the Debian distro applies for the Asterisk project, as per
ASTERISK-24191.

Review: https://reviewboard.asterisk.org/r/3896/

ASTERISK-24171 #close
Reported by: Jeremy Laine
patches:
  aelparse.8 uploaded by Jeremy Laine (License 6561)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422375 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds support for an <example /> tag in the XML documentation schema.

For CLI help, this doesn't change the formatting too much:
 - Preceeding white space is removed
 - Unlike with para elements, new lines are preserved

However, having an <example /> tag in the XML schema allows for the wiki
documentation generation script to surround the documentation with {code} or
{noformat} tags, generating much better content for the wiki - and allowing us
to put dialplan examples (and other code snippets, if desired) into the
documentation for an application/function/AMI command/etc.

Review: https://reviewboard.asterisk.org/r/3807/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419822 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Allow for responses to AMI actions/commands to be documented properly
in XML and displayed via the CLI. Response events are documented
exactly as standard AMI events are documented.

Review: https://reviewboard.asterisk.org/r/3812/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419342 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027

Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue.
Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample.

(issue ASTERISK-23061)
(issue ASTERISK-23028)
(issue ASTERISK-23046)
(issue ASTERISK-23027)
(closes issue ASTERISK-23061)
(closes issue ASTERISK-23028)
(closes issue ASTERISK-23046)
(closes issue ASTERISK-23027)
Reported by: Eugene, Jeremy Laine, Denis Pantsyrev
Patches:
 transferred.patch uploaded by Jeremy Laine (license 6561)
 hyphen.patch uploaded by Jeremy Laine (license 6561)
 sip.conf.sample.patch uploaded by Eugene (license 6360)
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Bridges have two new optional properties, a creator and a name.
Certain consumers of bridges will automatically provide bridges that
they create with these properties. Examples include app_bridgewait,
res_parking, app_confbridge, and app_agent_pool. In addition, a name
may now be provided as an argument to the POST function for creating
new bridges via ARI.

(closes issue AFS-47)
Review: https://reviewboard.asterisk.org/r/3070/
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404043 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a new function, PJSIP_ENDPOINT, which lets the dialplan query,
for any endpoint, any property configured on an endpoint. This function is a
companion to the CHANNEL function, which can be used to extract the endpoint
name for a channel.

Review: https://reviewboard.asterisk.org/r/3035
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403617 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/2898/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400282 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.

Review: https://reviewboard.asterisk.org/r/2708/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/2616

(closes issue ASTERISK-21542)
Reported by Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392729 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch allows a module to define its configuration in XML in source, such
that it can be parsed by the XML documentation engine. Documentation is
generated in a two-pass approach:

1. The documentation is first generated from the XML pulled from the source
2. The documentation is then enhanced by the registration of configuration
   options that use the configuration framework

This patch include configuration documentation for the following modules:
 * chan_motif
 * res_xmpp
 * app_confbridge
 * app_skel
 * udptl

Two new CLI commands have been added:
 * config show help - show configuration help by module, category, and item
 * xmldoc dump - dump the in-memory representation of the XML documentation to
   a new XML file.

Review: https://reviewboard.asterisk.org/r/2278
Review: https://reviewboard.asterisk.org/r/2058

patches:
  on review 2058 uploaded by twilson



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381527 65c4cc65-6c06-0410-ace0-fbb531ad65f3

file:///srv/subversion/repos/asterisk/trunk

................
  r379229 | mjordan | 2013-01-16 11:46:15 -0600 (Wed, 16 Jan 2013) | 10 lines
  
  Let documentation reference links specify which module they're linking to
  
  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.
  ........
  
  Merged revisions 379228 from http://svn.asterisk.org/svn/asterisk/branches/11
................
  r379231 | rmudgett | 2013-01-16 11:49:52 -0600 (Wed, 16 Jan 2013) | 10 lines
  
  chan_misdn: Fix compile error.
  
  (issue ASTERISK-15456)
  ........
  
  Merged revisions 379226 from http://svn.asterisk.org/svn/asterisk/branches/1.8
  ........
  
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................
  r379233 | rmudgett | 2013-01-16 12:09:28 -0600 (Wed, 16 Jan 2013) | 8 lines
  
  Reduce call-id logging resource usage.
  
  Since there is no need for the call-id logging ao2 object to have a lock,
  don't create it with one.
  ........
  
  Merged revisions 379232 from http://svn.asterisk.org/svn/asterisk/branches/11
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@379243 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379229 65c4cc65-6c06-0410-ace0-fbb531ad65f3

file:///srv/subversion/repos/asterisk/trunk

................
  r379211 | mjordan | 2013-01-16 09:33:05 -0600 (Wed, 16 Jan 2013) | 21 lines
  
  Multiple revisions 379209-379210
  
  ........
    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.
  ........
  
  Merged revisions 379209-379210 from http://svn.asterisk.org/svn/asterisk/branches/11
................


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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379211 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* 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)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371964 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370278 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369905 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369346 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/10

................
  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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https://origsvn.digium.com/svn/asterisk/branches/10

................
  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.
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331144 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/10

................
  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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https://origsvn.digium.com/svn/asterisk/branches/10

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331140 65c4cc65-6c06-0410-ace0-fbb531ad65f3