his patch implements the ARI API's for stored recordings. While the
original task only specified deleting a recording, it was simple
enough to implement the GET for all recordings, and for an individual
recording.

The recording playback operation was modified to use the same code for
accessing the recording as the REST API, so that they will behave
consistently.

There were several problems with the api-docs that were also fixed,
bringing the ARI spec in line with the implementation. There were some
'wishful thinking' fields on the stored recording model (duration and
timestamp) that were removed, because I ended up not implementing a
metadata file to go along with the recording to store such information.

The GET /recordings/live operation was removed, since it's not really
that useful to get a list of all recordings that are currently going
on in the system. (At least, if we did that, we'd probably want to
also list all of the current playbacks. Which seems weird.)

(closes issue ASTERISK-21582)
Review: https://reviewboard.asterisk.org/r/2693/
........

Merged revisions 397985 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397988 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:

* The word "Gulp" in dialstrings, functions, and CLI commands is now
  "PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This renames all files and API calls from several variants of
Stasis-HTTP to ARI including:
* Stasis-HTTP -> ARI
* STASIS_HTTP -> ARI
* stasis_http -> ari (ast_ari for global symbols, file names as well)
* stasis http -> ARI

Review: https://reviewboard.asterisk.org/r/2706/
(closes issue ASTERISK-22136)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395603 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds authentication support to ARI.

Two authentication methods are supported. The first is HTTP Basic
authentication, as specified in RFC 2617[1]. The second is by simply
passing the username and password as an ?api_key query parameter
(which allows swagger-ui[2] to authenticate more easily).

ARI usernames and passwords are configured in the ari.conf file
(formerly known as stasis_http.conf). The user may be set to
`read_only`, which will prohibit the user from issuing POST, DELETE,
etc. Also, the user's password may be specified in either plaintext,
or encrypted using the crypt() function.

Several other notes about the patch.

 * A few command line commands for seeing ARI config and status were
   also added.
 * The configuration parsing grew big enough that I extracted it to
   its own file.

 [1]: http://www.ietf.org/rfc/rfc2617.txt [2]:
 https://github.com/wordnik/swagger-ui

(closes issue ASTERISK-21277)
Review: https://reviewboard.asterisk.org/r/2649/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393530 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch started with the simple idea of changing the /events data
model to be more sane. The original model would send out events like:

    { "stasis_start": { "args": [], "channel": { ... } } }

The event discriminator was the field name instead of being a value in
the object, due to limitations in how Swagger 1.1 could model objects.
While technically sufficient in communicating event information, it was
really difficult to deal with in terms of client side JSON handling.

This patch takes advantage of a proposed extension[1] to Swagger which
allows type variance through the use of a discriminator field. This had
a domino effect that made this a surprisingly large patch.

 [1]: https://groups.google.com/d/msg/wordnik-api/EC3rGajE0os/ey_5dBI_jWcJ

In changing the models, I also had to change the swagger_model.py
processor so it can handle the type discriminator and subtyping. I took
that a big step forward, and using that information to generate an
ari_model module, which can validate a JSON object against the Swagger
model.

The REST and WebSocket generators were changed to take advantage of the
validators. If compiled with AST_DEVMODE enabled, JSON objects that
don't match their corresponding models will not be sent out. For REST
API calls, a 500 Internal Server response is sent. For WebSockets, the
invalid JSON message is replaced with an error message.

Since this took over about half of the job of the existing JSON
generators, and the .to_json virtual function on messages took over the
other half, I reluctantly removed the generators.

The validators turned up all sorts of errors and inconsistencies in our
data models, and the code. These were cleaned up, with checks in the
code generator avoid some of the consistency problems in the future.

 * The model for a channel snapshot was trimmed down to match the
   information sent via AMI. Many of the field being sent were not
   useful in the general case.
 * The model for a bridge snapshot was updated to be more consistent
   with the other ARI models.

Another impact of introducing subtyping was that the swagger-codegen
documentation generator was insufficient (at least until it catches up
with Swagger 1.2). I wanted it to be easier to generate docs for the API
anyways, so I ported the wiki pages to use the Asterisk Swagger
generator. In the process, I was able to clean up many of the model
links, which would occasionally give inconsistent results on the wiki. I
also added error responses to the wiki docs, making the wiki
documentation more complete.

Finally, since Stasis-HTTP will now be named Asterisk REST Interface
(ARI), any new functions and files I created carry the ari_ prefix. I
changed a few stasis_http references to ari where it was non-intrusive
and made sense.

(closes issue ASTERISK-21885)
Review: https://reviewboard.asterisk.org/r/2639/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393529 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch moves the RESTful URL's around to more appropriate
locations for release.

The /stasis URL's are moved to /ari, since Asterisk REST Interface was
a more appropriate name than Stasis-HTTP. (Most of the code still has
stasis_http references, but they will be cleaned up after there are no
more outstanding branches that would have merge conflicts with such a
change).

A larger change was moving the ARI events WebSocket off of the shared
/ws URL to its permanent home on /ari/events. The Swagger code
generator was extended to handle "upgrade: websocket" and
"websocketProtocol:" attributes on an operation.

The WebSocket module was modified to better handle WebSocket servers
that have a single registered protocol handler. If a client
connections does not specify the Sec-WebSocket-Protocol header, and
the server has a single protocol handler registered, the WebSocket
server will go ahead and accept the client for that subprotocol.

(closes issue ASTERISK-21857)
Review: https://reviewboard.asterisk.org/r/2621/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393528 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Breaks many things until they can be reworked.  A partial list:
chan_agent
chan_dahdi, chan_misdn, chan_iax2 native bridging
app_queue
COLP updates
DTMF attended transfers
Protocol attended transfers


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When implementing playback for stasis-http, the monolithicedness of
res_stasis really started to get in my way.

This patch breaks the major components of res_stasis.c into individual
files.

 * res/stasis/app.c - Stasis application tracking
 * res/stasis/control.c - Channel control objects
 * res/stasis/command.c - Channel command object

This refactoring also allows res_stasis applications to be loaded as
independent modules, such as the new res_stasis_answer module.

The bulk of this patch is simply moving code from one file to another,
adjusting names and adding accessors as necessary.

Review: https://reviewboard.asterisk.org/r/2530/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388729 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The pimp_my_sip branch is being merged at this point because
it offers basic functionality, and from an API standpoint, things
are complete.

SIP work is *not* feature-complete; however, with the completion
of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have
been created, and thus it is possible for developers to attempt
to create new SIP work.

API documentation can be found in the doxygen in the code, but
usability documentation is still lacking.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The API itself is documented using Swagger, a lightweight mechanism for
documenting RESTful API's using JSON. This allows us to use swagger-ui
to provide executable documentation for the API, generate client
bindings in different languages, and generate a lot of the boilerplate
code for implementing the RESTful bindings. The API docs live in the
rest-api/ directory.

The RESTful bindings are generated from the Swagger API docs using a set
of Mustache templates.  The code generator is written in Python, and
uses Pystache. Pystache has no dependencies, and be installed easily
using pip. Code generation code lives in rest-api-templates/.

The generated code reduces a lot of boilerplate when it comes to
handling HTTP requests. It also helps us have greater consistency in the
REST API.

(closes issue ASTERISK-20891)
Review: https://reviewboard.asterisk.org/r/2376/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386232 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ICE/STUN/TURN support in res_rtp_asterisk is also now optional.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382900 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Allow parallel builds, better tolerate failures, build faster.

This also stops running dependencies before top-level configure has been run.

(closes issue ASTERISK-20815)

Review: https://reviewboard.asterisk.org/r/2292/
........

Merged revisions 380816 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380817 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Update and extend the configuration_file group and enable linking to the resource.  Update title that was left behind many years ago.

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375003 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixes a build regression introduced in r369517 "Add support for ICE/STUN/TURN
in res_rtp_asterisk and chan_sip." [1].

[1] http://svnview.digium.com/svn/asterisk?view=revision&revision=369517

When compiling asterisk in parallel like:
    $ make -j 10

It's possible to get errors like the following:

    .pjlib-util-test-x86_64-unknown-linux-gnu.depend:120: *** missing separator.  Stop.
    make[4]: *** [depend] Error 2
    make[3]: *** [dep] Error 1
    make[2]: *** [/home/sruffell/asterisk-working/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a] Error 2
    make[3]: warning: jobserver unavailable: using -j1.  Add `+' to parent make rule.

This is because the build system is trying to build each of the libraries in
pjproject in parallel. Now the build will build pjproject in a single job and
link the results into res_asterisk_rtp.

Parallel builds, on one test system, saves ~1.5 minutes from a default Asterisk
build:

Single job:
    $ git clean -fdx >/dev/null && time ( ./configure >/dev/null 2>&1 && make >/dev/null 2>&1 )

    real    2m34.529s
    user    1m41.810s
    sys     0m15.970s

Parallel make:
    $ git clean -fdx >/dev/null && time ( ./configure >/dev/null 2>&1 && make -j10 >/dev/null 2>&1 )

    real    1m2.353s
    user    2m39.120s
    sys     0m18.850s

(closes issue ASTERISK-20362)
Reported by: Shaun Ruffel
Patches:
    0001-res_asterisk_rtp-Fix-build-error-when-using-parallel.patch uploaded by Shaun Ruffel (License #5417)
........

Merged revisions 372609 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372610 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370240 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Turn on debugging for pjproject so we can get a better idea of what is causing the generic CCSS test crash.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370234 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/1891/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369517 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318351 | rmudgett | 2011-05-09 18:15:32 -0500 (Mon, 09 May 2011) | 6 lines
  
  Remove references to res_features and its export file.
  
  The contents of res/res_features.c was moved to into main/features.c
  awhile ago.  There is no longer any need for the res/Makefile to reference
  res_features or the res_features linker exports file to exist.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318352 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r262421 | qwell | 2010-05-11 14:55:42 -0500 (Tue, 11 May 2010) | 11 lines
  
  Use a less silly method for modifying a flex-generated file.
  
  The sed syntax that was used wasn't actually valid, causing some versions to
  choke.  This is the method that is used in 1.6.x+ for similar changes.
  
  (closes issue #16696)
  Reported by: bklang
  Patches: 
        16696-sedfix.diff uploaded by qwell (license 4)
  Tested by: qwell
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262422 65c4cc65-6c06-0410-ace0-fbb531ad65f3

users expect them to work.

'core set debug' and 'core set verbose' can optionally change the
level for a specific filename; however, this is actually for a
specific source file name, not the module that source file is included
in. With examples like chan_sip, chan_iax2, chan_misdn and others
consisting of multiple source files, this will not lead to the
behavior that users expect. If they want to set the debug level for
chan_sip, they want it set for all of chan_sip, and not to have to
also set it for reqresp_parser and other files that comprise the
chan_sip module.

This patch changes this functionality to be module-name based instead
of file-name based.

To make this work, some Makefile modifications were required to ensure
that the AST_MODULE definition is present in each object file produced
for each module as well.

Review: https://reviewboard.asterisk.org/r/574/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253917 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r242969 | tilghman | 2010-01-25 15:50:22 -0600 (Mon, 25 Jan 2010) | 2 lines
  
  Err, and use the new menuselect define, too.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242971 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r242852 | tilghman | 2010-01-25 14:15:45 -0600 (Mon, 25 Jan 2010) | 2 lines
  
  Restore FreeBSD to able-to-compile-ish-mode
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242857 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r242728 | tilghman | 2010-01-24 23:42:22 -0600 (Sun, 24 Jan 2010) | 2 lines
  
  Buildbot pointed out an error (thanks, buildbot!)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242729 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r242723 | tilghman | 2010-01-24 23:33:37 -0600 (Sun, 24 Jan 2010) | 2 lines
  
  Oops, should have used CMD_PREFIX, not ECHO_PREFIX, for the commands.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242724 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010) | 8 lines
  
  Only rebuild bison and flex source files on demand, if bison and flex are detected by the configure script.
  
  Changed after discussion on the -dev list about possible unnecessary build
  failures, due to checkouts/untars causing these special source files to
  possibly be newer than their resulting C files.  This should additionally
  ensure that nobody need learn about extra Makefile arguments to ensure the
  proper files get rebuilt when changes are made to these special source files.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242521 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r242423 | tilghman | 2010-01-22 15:44:18 -0600 (Fri, 22 Jan 2010) | 7 lines
  
  Rebuild from flex, bison sources when necessary.
  
  (issue #14629)
   Reported by: Marquis
   Patches: 
         20100121__issue14629.diff.txt uploaded by tilghman (license 14)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242424 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines
  
  Ensure that user-provided CFLAGS and LDFLAGS are honored.
  
  This commit changes the build system so that user-provided flags (in ASTCFLAGS
  and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
  by the build system itself, so that the user can effectively override the
  build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
  be provided *either* in the environment before running 'make', or as variable
  assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
  is no longer necessary, so they are no longer documented, but are still supported
  so as not to break existing build systems that supply them when building Asterisk.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207680 65c4cc65-6c06-0410-ace0-fbb531ad65f3

They removed the LDAP_DEPRECATED define from their source and since we are using a couple
of deprecated function calls we should define it with a CFLAG.

Tested by me on OpenBSD 4.4 and snuff-home on Linux to make sure everything keeps compiling.
It shouldn't break, we only define the LDAP_DEPRECATED with this which is what 
all 2.2.X and older versions of OpenLDAP did in their own tree.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159734 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines
  
  the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems.
  
  with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course).
  
  while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157974 65c4cc65-6c06-0410-ace0-fbb531ad65f3

To make sure nobody commits script-modified files we first make a backup
of asterisk.tex, run the script, generate the pdf and / or html,
and put the original asterisk.tex back.
This will guard us for the stuff that happened before that someone committed 
a locally modified asterisk.tex, with changes done by this script.

(closes issue #13062)
Reported by: mvanbaak
Patches:
      sed_without-i-v3.diff uploaded by mvanbaak (license 7)
Tested by: mvanbaak

Feedback from Corydon. Thanks for taking the time to go through this.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130578 65c4cc65-6c06-0410-ace0-fbb531ad65f3

make the AIS checking a little more generic, and have a more useful configure script command line option for OpenAIS


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127017 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This commit merges in the rest of the code needed to support distributed device
state.  There are two main parts to this commit.

Core changes:
 - The device state handling in the core has been updated to understand device
   state across a cluster of Asterisk servers.  Every time the state of a device
   changes, it looks at all of the device states on each node, and determines the
   aggregate device state.  That resulting device state is what is provided to
   modules in Asterisk that take actions based on the state of a device.

New module, res_ais:
 - A module has been written to facilitate the communication of events between
   nodes in a cluster of Asterisk servers.  This module uses the SAForum AIS
   (Service Availability Forum Application Interface Specification) CLM and EVT
   services (Cluster Management and Event) to handle this task.  This module
   currently supports sharing Voicemail MWI (Message Waiting Indication) and
   device state events between servers.  It has been tested with openais, though
   other implementations of the spec do exist.

For more information on testing distributed device state, see the following doc:
  - doc/distributed_devstate.txt


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121559 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar 2008) | 11 lines

fix up various compiler warnings found with gcc-4.3:

- the output of flex includes a static function called 'input' that is not used, so for the moment we'll stop having the compiler tell us about unused variables in the flex source files (a better fix would be to improve our flex post-processing to remove the unused function)

- main/stdtime/localtime.c makes assumptions about signed integer overflow, and gcc-4.3's improved optimizer tries to take advantage of handling potential overflow conditions at compile time; for now, suppress these optimizations until we can fiure out if the code needs improvement

- main/udptl.c has some references to uninitialized variables; in one case there was no bug, but in the other it was certainly possibly for unexpected behavior to occur

- main/editline/readline.c had an unused variable


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107373 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r97889 | murf | 2008-01-10 14:37:10 -0700 (Thu, 10 Jan 2008) | 1 line

Applied the same fixes for ael.flex as was done in 97849 for ast_expr2.fl; overrode the normally generate yyfree func with our own version that checks the pointer for non-null before passing to free(). Also takes care of a little problem with 2.5.33 and the use of the __STDC_VERSION__ macro.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97890 65c4cc65-6c06-0410-ace0-fbb531ad65f3

go back to including libresample in the main Asterisk binary, but this time including a small hack to ensure that it does get linked in (and also modify the strip_nonapi script to leave the resample_<foo> symbols alone)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95816 65c4cc65-6c06-0410-ace0-fbb531ad65f3

res_resample, and mark codec_resample as dependent upon res_resample.  This
prevents the linker from optimizing away libresample, and also makes it so the
libresample code isn't linked in to multiple places.  (I have another module
in a branch that needs it, too.)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95697 65c4cc65-6c06-0410-ace0-fbb531ad65f3

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r93180 | kpfleming | 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007) | 23 lines

In http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
rizzo brought up some issues related to the way that the metadata required
for menuselect and the rest of the build system is extracted from the source
files. Since I had a few hours to kill on an airplane today, I decided to
improve this situation... so now the system caches the extracted metadata
and uses it to build the menuselect 'tree' as much as it can. The result
of this is that when a single source file is changed, only the metadata for
that file needs to be extracted again, and the rest is used from the cache
files. I also reduced the number of forked processes required to do the
metadata extraction; it was actually possible to do most of what we needed
in the Makefiles themselves without using any shell scripts at all! On my
laptop, these changes resulted in an 80% decrease in the time required
for the 'menuselect.makeopts' automatic check to occur after editing a single
source file.

While doing this work I also cleaned up a few minor things in the Makefiles,
adding a check for 'awk' to the configure script and changed all remaining
places we use 'grep' or 'awk' to use the ones found by the configure script,
and changed the 'prep_tarball' script to build the menuselect metadata so
that tarballs of Asterisk will include it and won't require the user to
wait while it is extracted after unpacking.


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93184 65c4cc65-6c06-0410-ace0-fbb531ad65f3

generate loadable and embedded module lists.

Individual Makefiles now are a lot simpler, possibly as simple as this:

    -include $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
    MODULE_PREFIX=cdr_
    all: _all
    include $(ASTTOPDIR)/Makefile.moddir_rules

and also more flexible because in a single directory we can combine
various types of modules (app_, cdr_, func_, ... ) by simply
listing them in the MODULE_PREFIX variable.

The individual Makefiles can also create list of modules to be
excluded by listing them in the variablel MODULE_EXCLUDE (see an
example in channels/Makefile).

With this change it becomes trivial to integrate a directory with
locally created/modified sources into the main build.




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92082 65c4cc65-6c06-0410-ace0-fbb531ad65f3

the top level directory.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92022 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90834 65c4cc65-6c06-0410-ace0-fbb531ad65f3