kpfleming: Author Summary

Make COMPILE_DOUBLE magic actually work.

The build system has some special magic to ensure that if Asterisk is built
with --enable-dev-mode *and* DONT_OPTIMIZE, that all the source is still compiled
with the optimizer enabled (even though the result will be thrown away), because
the compiler is able to find a great deal of coding errors and bugs as a result
of running its optimizers. Unfortunately at some point this mode got broken,
and the 'throwaway' compile of the code was no longer done with the optimizer
enabled. This patch corrects that problem.
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Merged revisions 357212 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Make COMPILE_DOUBLE magic actually work.

The build system has some special magic to ensure that if Asterisk is built
with --enable-dev-mode *and* DONT_OPTIMIZE, that all the source is still compiled
with the optimizer enabled (even though the result will be thrown away), because
the compiler is able to find a great deal of coding errors and bugs as a result
of running its optimizers. Unfortunately at some point this mode got broken,
and the 'throwaway' compile of the code was no longer done with the optimizer
enabled. This patch corrects that problem.

Address OpenSSL initialization issues when using third-party libraries.

When Asterisk is used with various third-party libraries (CURL, PostgresSQL,
many others) that have the ability themselves to use OpenSSL, it is possible
for conflicts to arise in how the OpenSSL libraries are initialized and
shutdown. This patch addresses these conflicts by 'wrapping' the important
functions from the OpenSSL libraries in a new shared library that is part
of Asterisk itself, and is loaded in such a way as to ensure that *all*
calls to these functions will be dispatched through the Asterisk wrapper
functions, not the native functions.

This new library is optional, but enabled by default. See the CHANGES file
for documentation on how to disable it.

Along the way, this patch also makes a few other minor changes:

* Changes MODULES_DIR to ASTMODDIR throughout the build system, in order to
  more closely match what is used during run-time configuration.

* Corrects some errors in the configure script where AC_CHECK_TOOLS was used
  instead of AC_PATH_PROG.

* Adds a new variable for linker flags in the build system (DYLINK), used for
  producing true shared libraries (as opposed to the dynamically loadable
  modules that the build system produces for 'regular' Asterisk modules).

* Moves the Makefile bits that handle installation and uninstallation of the
  main Asterisk binary into main/Makefile from the top-level Makefile.

* Moves a couple of useful preprocessor macros from optional_api.h to
  asterisk.h.

Review: https://reviewboard.asterisk.org/r/1006/

Clarify log WARNING message when port-zero SDP 'm' lines received.

Previously, if an m-line in an SDP offer or answer had a port number of zero,
that line was skipped, and resulted in an 'Unsupported SDP media type...'
warning message. This was misleading, as the media type was not unsupported,
but was ignored because the m-line indicated that the media stream had been
rejected (in an answer) or was not going to be used (in an offer).
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Merged revisions 353260 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 353261 from http://svn.asterisk.org/svn/asterisk/branches/10

Clarify log WARNING message when port-zero SDP 'm' lines received.

Previously, if an m-line in an SDP offer or answer had a port number of zero,
that line was skipped, and resulted in an 'Unsupported SDP media type...'
warning message. This was misleading, as the media type was not unsupported,
but was ignored because the m-line indicated that the media stream had been
rejected (in an answer) or was not going to be used (in an offer).
........

Merged revisions 353260 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Add 'L16-256' MIME subtype alias for slin16.

Asterisk has supported the 'L16' MIME subtype for 16kHz signed linear (PCM)
audio for quite some time, but some endpoints refer to it as 'L16-256'. This
commit adds this as an alias for the existing format.
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Merged revisions 353126 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 353127 from http://svn.asterisk.org/svn/asterisk/branches/10

Add 'L16-256' MIME subtype alias for slin16.

Asterisk has supported the 'L16' MIME subtype for 16kHz signed linear (PCM)
audio for quite some time, but some endpoints refer to it as 'L16-256'. This
commit adds this as an alias for the existing format.
........

Merged revisions 353126 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Remove "asterisk/version.h" in favor of "asterisk/ast_version.h".

A long time ago, in a land far far away, we added "asterisk/ast_version.h",
which provides the ast_get_version() and ast_get_version_num() functions. These
were added so that modules that needed the version information for the Asterisk
instance they were loaded in could actually get it (as opposed the version that
they were compiled against). We changed everything in the tree to use the
new mechanism (although later main/test.c was added using the old method).
However, the old mechanism was never removed, and as a result, new code is
still trying to use it.

This commit removes asterisk/version.h and replaces it with a header that
will generate a compile-time error if you try to use it (the error message
tells you which header you should use instead). It also removes the Makefile
and build_tools bits that generated the file, and it updates main/test.c to
use the 'proper' method of getting the Asterisk version information.

This is an API change and thus is being committed for trunk only, but it's
a fairly minor one and definitely improves the situation for out-of-tree
modules.

Blocked revisions 352616

........
Avoid unnecessary rebuilds of main/test.c.

main/test.c includes "asterisk/version.h", when it should include
"asterisk/ast_version.h" instead (and it should use the ast_get_version()
and ast_get_version_num() functions). This commit modifies it to extract
the Asterisk version information using the proper APIs, and as a result means
that main/test.c no longer needs to be rebuilt when a Subversion checkout
is updated or modified.
........

Merged revisions 352612 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Avoid unnecessary rebuilds of main/test.c.

main/test.c includes "asterisk/version.h", when it should include
"asterisk/ast_version.h" instead (and it should use the ast_get_version()
and ast_get_version_num() functions). This commit modifies it to extract
the Asterisk version information using the proper APIs, and as a result means
that main/test.c no longer needs to be rebuilt when a Subversion checkout
is updated or modified.
........

Merged revisions 352612 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Address OpenSSL initialization issues when using third-party libraries.

When Asterisk is used with various third-party libraries (CURL, PostgresSQL,
many others) that have the ability themselves to use OpenSSL, it is possible
for conflicts to arise in how the OpenSSL libraries are initialized and
shutdown. This patch addresses these conflicts by 'wrapping' the important
functions from the OpenSSL libraries in a new shared library that is part
of Asterisk itself, and is loaded in such a way as to ensure that *all*
calls to these functions will be dispatched through the Asterisk wrapper
functions, not the native functions.

This new library is optional, but enabled by default. See the CHANGES file
for documentation on how to disable it.

Along the way, this patch also makes a few other minor changes:

* Changes MODULES_DIR to ASTMODDIR throughout the build system, in order to
  more closely match what is used during run-time configuration.

* Corrects some errors in the configure script where AC_CHECK_TOOLS was used
  instead of AC_PATH_PROG.

* Adds a new variable for linker flags in the build system (DYLINK), used for
  producing true shared libraries (as opposed to the dynamically loadable
  modules that the build system produces for 'regular' Asterisk modules).

* Moves the Makefile bits that handle installation and uninstallation of the
  main Asterisk binary into main/Makefile from the top-level Makefile.

* Moves a couple of useful preprocessor macros from optional_api.h to
  asterisk.h.

Review: https://reviewboard.asterisk.org/r/1006/

Remove "asterisk/version.h" in favor of "asterisk/ast_version.h".

A long time ago, in a land far far away, we added "asterisk/ast_version.h",
which provides the ast_get_version() and ast_get_version_num() functions. These
were added so that modules that needed the version information for the Asterisk
instance they were loaded in could actually get it (as opposed the version that
they were compiled against). We changed everything in the tree to use the
new mechanism (although later main/test.c was added using the old method).
However, the old mechanism was never removed, and as a result, new code is
still trying to use it.

This commit removes asterisk/version.h and replaces it with a header that
will generate a compile-time error if you try to use it (the error message
tells you which header you should use instead). It also removes the Makefile
and build_tools bits that generated the file, and it updates main/test.c to
use the 'proper' method of getting the Asterisk version information.

This is an API change and thus is being committed for trunk only, but it's
a fairly minor one and definitely improves the situation for out-of-tree
modules.

Correct two flaws in sip.conf.sample related to AST-2011-013.

* The sample file listed *two* values for the 'nat' option as being the default.
  Only 'force_rport' is the default.

* The warning about having differing 'nat' settings confusingly referred to both
  peers and users.
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Merged revisions 348515 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
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Merged revisions 348516 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 348517 from http://svn.asterisk.org/svn/asterisk/branches/10

Modify comments in MeetMe application documentation about DAHDI.

The MeetMe application documentation has some comments about usage of DAHDI,
and they were a bit outdated relative to modern DAHDI releases. This patch
changes the comment to just tell the user that a functional DAHDI timing
source is required, and no longer mention 'dahdi_dummy', since that module
does not exist in current DAHDI releases.
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Merged revisions 342990 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 342991 from http://svn.asterisk.org/svn/asterisk/branches/10

Modify comments in MeetMe application documentation about DAHDI.

The MeetMe application documentation has some comments about usage of DAHDI,
and they were a bit outdated relative to modern DAHDI releases. This patch
changes the comment to just tell the user that a functional DAHDI timing
source is required, and no longer mention 'dahdi_dummy', since that module
does not exist in current DAHDI releases.
........

Merged revisions 342990 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Ensure that support-level separator items are skipped in every process that
iterates over members of categories in the menuselect tree.

Merged revisions 328879 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/2.0

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  r328879 | kpfleming | 2011-07-19 16:31:16 -0500 (Tue, 19 Jul 2011) | 23 lines
  
  Merged revisions 328878 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r328878 | kpfleming | 2011-07-19 16:29:07 -0500 (Tue, 19 Jul 2011) | 17 lines
    
    Revert partial attempt at handling pathnames with spaces.
    
    Revision 299794 attempted to improve the build system to be able to handle
    pathnames (primarily DESTDIR) with spaces in them, since this is common on
    some platforms (including Mac OSX). Unfortunately, the changes were incomplete
    and did not actually provide the desired behavior, and as a side effect the
    functionality that ensured that stale headers in the Asterisk 'include' directory
    were removed got broken. In addition, the check for stale (and possibly
    incompatible) modules in the Asterisk 'modules' directory also got broken, and
    would never report any stale modules. Users upgrading to this version or later
    versions would then see unexpected module load errors.
    
    Since there are few users who actually want to install Asterisk into paths
    that contain spaces, and a proper fix for the build system would take many hours,
    the best solution for now is to just revert the partial solution.
  ........
................

Edit the merge properties to match their names.

Merged revisions 325416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r325416 | kpfleming | 2011-06-28 16:50:43 -0500 (Tue, 28 Jun 2011) | 3 lines
  
  Fix random misspelling noticed on asterisk-users.
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Merged revisions 320560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r320560 | kpfleming | 2011-05-23 10:47:14 -0500 (Mon, 23 May 2011) | 4 lines
  
  Don't generate spurious "No: command not found" messages when running the
  configure script on a system that has neither gmime-config nor pkg-config.
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Fix bug with 'F' option for ReceiveFAX and SendFAX.

Skipping the call to set_t38_fax_caps() caused the FAX session
details to not be marked as supporting audio FAX either... the
function's name is a bit misleading. This patch restores the
single bit of non-T.38 behavior from that function when audio
mode is forced.

Rename the SendFAX/ReceiveFAX 'force audio' option.

The recently added option to disable usage of T.38 for a single
session should have been named 'F' for 'force audio', since that
is really what the user is asking to happen (and it's a positive
option instead of a negative option that way).

Add ability to disable T.38 usage for specific SendFAX/ReceiveFAX sessions.

Sometimes during troubleshooting it can be useful to disable T.38 usage in order
to narrow down a problem. This patch adds an 'n' option to SendFAX and ReceiveFAX
so that can be done without having to disable T.38 usage entirely for the peer
that Asterisk is communicating with.

(inspired by trying to assist Bryant Zimmerman on asterisk-users)

Avoid unnecessary rebuilds of main/test.c.

main/test.c includes "asterisk/version.h", when it should include
"asterisk/ast_version.h" instead (and it should use the ast_get_version()
and ast_get_version_num() functions). This commit modifies it to extract
the Asterisk version information using the proper APIs, and as a result means
that main/test.c no longer needs to be rebuilt when a Subversion checkout
is updated or modified.
........

Merged revisions 352612 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Change the internal name of the menuselect options that are used to control
whether modules are embedded or not; using just the bare category name led to
accidentally enabling these options when users used the wrong "--enable"
operation on the menuselect command line.

Now the internal option names are prefixed with "EMBED_", so they won't be
the same as the name of the category containing the modules they control
the embedding of.
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Merged revisions 341022 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 341023 from http://svn.asterisk.org/svn/asterisk/branches/10

Kill red blobs with fire! (eliminate trailing whitespace)

Entries without a support level should be listed as 'unspecified', to keep
users from wondering what 'other' support level has been assigned to them.

Ensure that support-level separator items are skipped in every process that
iterates over members of categories in the menuselect tree.

Code cleanup:

* Properly use 'enum support_level_values' type in code that uses these values,
  rather than 'int'.

* Ensure that all member items in categories have a non-empty support_level,
  "unspecified" unless it is overridden by the input files.

* Don't show a support level of 'core' in the user interface code if a member
  does not have a support level (see above... they will always have one now).

Code cleanup:

* Properly use 'enum support_level_values' type in code that uses these values,
  rather than 'int'.

* Ensure that all member items in categories have a non-empty support_level,
  "unspecified" unless it is overridden by the input files.

* Don't show a support level of 'core' in the user interface code if a member
  does not have a support level (see above... they will always have one now).

Code cleanup:

* Properly use 'enum support_level_values' type in code that uses these values,
  rather than 'int'.

* Ensure that all member items in categories have a non-empty support_level,
  "unspecified" unless it is overridden by the input files.

* Don't show a support level of 'core' in the user interface code if a member
  does not have a support level (see above... they will always have one now).

Kill red blobs with fire! (eliminate trailing whitespace)

Merged revisions 325416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r325416 | kpfleming | 2011-06-28 16:50:43 -0500 (Tue, 28 Jun 2011) | 3 lines
  
  Fix random misspelling noticed on asterisk-users.
........