Commit Graph

11 Commits

Author SHA1 Message Date
Matthew Jordan d04ab3c645 Add CLI configuration documentation
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
2013-02-15 13:38:12 +00:00
Kevin P. Fleming 166b4e2b30 Multiple revisions 369001-369002
........
  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
2012-06-15 16:20:16 +00:00
Eliel C. Sardanons a753e8878b Asterisk data retrieval API.
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/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-22 18:07:02 +00:00
Mark Michelson e24661fd18 Merge Call completion support into trunk.
From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:

1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
   multiple calls to the same device. This proved to not be such a good idea
   when implementing protocol-specific monitors, and so we ended up using one
   monitor per-device per-call.
3. There are some configuration options which were conceived after the document
   was written. These are documented in the ccss.conf.sample that is on this
   review request.
		      
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.

This implements CCBS and CCNR in several flavors.

First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.

Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:

* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
  what is defined in the referenced draft.

* Implementation of the draft required support for SIP PUBLISH. I attempted to write
  this in a generic-enough fashion such that if someone were to want to write PUBLISH
  support for other event packages, such as dialog-state or presence, most of the effort
  would be in writing callbacks specific to the event package.

* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
  parser. The PIDF support added is a bit minimal. I first wrote a validation
  routine to ensure that the PIDF document is formatted properly. The rest of the
  PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
  code. In other words, while there is PIDF support here, it is not in any state
  where it could easily be applied to other event packages as is.

Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.

Review: https://reviewboard.asterisk.org/r/523


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09 15:31:32 +00:00
Kevin P. Fleming 4222e1c367 Define our desires for POSIX and X/OPEN API features properly.
Based on a post on the gcc-help mailing list and some subsequent reading,
we can increase our portability to various platforms by directly defining
the POSIX and X/OPEN API feature sets we wish to have available. This patch
does that, and also includes a double-check to ensure that the system
we are compiling on can actually provide the requested feature sets.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@212463 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-17 16:25:10 +00:00
Tilghman Lesher ec99151d74 Revert commit 192032. This define is needed on Mac OS X.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-27 23:48:15 +00:00
Eliel C. Sardanons a007eb13b2 Do not avoid loading the XML documentation if not XInclude substitution is done.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18 13:30:34 +00:00
Eliel C. Sardanons 2f996a3ab4 Allow to include sections of other parts of the xml documentation.
Avoid duplicating xml documentation by allowing to include other parts of
the xml documentation using XInclude.
Example:
   <xi:include xpointer="xpointer(/docs/function[@name='CHANNEL']/synopsis)" />
(Insert this line to include the synopsis of the CHANNEL function xml
documentation).

It is also possible to include documentation from other files in the
'documentation/' directory using the href="" attribute inside a xinclude
element.

(closes issue #15107)
Reported by: lmadsen

(issue #14444)
Reported by: ewieling



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194982 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-16 20:01:22 +00:00
Eliel C. Sardanons 150edbb67d Do not re-define _POSIX_C_SOURCE if it was already defined.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-04 15:35:35 +00:00
Tilghman Lesher e179e613f7 1. Make OS X compile cleanly with app_stack.
2. Use curl to download sound files, as curl is installed natively on OS X,
whereas wget and fetch are not.
(closes issue #14332)
 Reported by: oej
 Tested by: Corydon76


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173130 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-03 00:29:49 +00:00
Russell Bryant 5b168ee34b Merge changes from team/group/appdocsxml
This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format.  Currently, a new format is available for
applications and dialplan functions.  A good number of conversions to the new format
are also included.

For more information, see the following message to asterisk-dev:

http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-01 21:10:07 +00:00