Changes made to apps/Makefile to optionally build all three app_voicemail
variations at the same time: 1) file (default), 2) odbc, and 3) imap.
This functionality was requested by users. modules.conf.sample warns the
user to make sure only one voicemail is loaded at a time.

Change-Id: Iba3cd8ffb4b7e8b1c64a11dd383e1eafcd3ed0e7

This officially deprecates chan_sip in Asterisk 17+.  A warning is
printed upon startup or module load to tell users that they should
consider migrating.  chan_sip is still built by default but the default
modules.conf skips loading it at startup.

Very important to note we are not scheduling a time where chan_sip will
be removed.  The goal of this change is to accurately inform end users
of the current state of chan_sip and encourage movement to the fully
supported chan_pjsip.

Change-Id: Icebd8848f63feab94ef882d36b2e99d73155af93

Change-Id: Id449d4435c38148b56ac4cfd61ae4d90ac66bb90

Change disables loading of res_hep.so in default installation.  Loading
res_hep has a performance impact whether it's used or not.  This disables
loading of it in sample config files.

Change-Id: I5ec150cf941634fabc72973e5bf1a965cb0ef9d0

The sample modules.conf explicitly loaded res_musiconhold.so.  This is
redundent as autoload=yes is already set.  It causes warnings if
res_musiconhold.so was not installed and results in an unexpected load
if the admin disables autoload without remembering to remove the
res_musiconhold load statement.

Also remove reference to unknown module pbx_gtkconsole.

Change-Id: Ib01888994d9f1364b14d3c9fb6ff96774a6e580a

This moves all samples configs from configs/ to configs/samples. This allows
for additional sets of sample configuration files to be added in the future.

Review: https://reviewboard.asterisk.org/r/3804/


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Review: https://reviewboard.asterisk.org/r/426/


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Let's try that again, this time removing trailing whitespace and not leading
whitespace.  I can't believe no one noticed.


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This configuration file was changed to ensure that only one console channel driver
(chan_oss) is loaded by default, but the change would only work if chan_console
was not built. Now it will work as expected; if chan_alsa or chan_console are built
and installed, they will not be loaded unless explicity requested.



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

........
r97753 | russell | 2008-01-10 10:19:47 -0600 (Thu, 10 Jan 2008) | 2 lines

Remove other remnants of pbx_kdeconsole

........


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based on configuration templates that use Asterisk dialplan function and
variable substitution.  It should be possible to create phone profiles and
templates that work for the majority of phones provisioned over http. It
is currently only intended to provision a single user account per phone.
An example profile and set of templates for Polycom phones is provided.
NOTE: Polycom firmware is not included, but should be placed in
AST_DATA_DIR/phoneprov/configs to match up with the included templates.



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Add a new console channel driver, chan_console, which is a console channel
driver that uses portaudio as a cross platform audio interface.  It was written
to provide a console channel driver that works with Mac CoreAudio, but it
supports a number of other audio interfaces, as well, including OSS and ALSA.
It could one day be the single console channel driver, but does not yet have
as many features as chan_oss.


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- Remove obsolete modules from modules.conf.sample
  (make install will warn if those exist on the machine)


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Note that the res_speech module will need to be loaded first, and add a conveient line to uncomment to do so for the time being.



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add support for 'early loading' modules, so that nearly all configuration files can be read from Realtime storage
add warning for when file mapping is found but the engine is not available
add warning for trying to map 'logger.conf', since it cannot be reliably mapped


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