Review: https://reviewboard.asterisk.org/r/1707/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353685 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The CEL eventtype field for ODBC and PGSQL backends should be USER_DEFINED
instead of the user defined event name supplied by the CELGenUserEvent
application.  If the field is output as a number, the user defined name
does not have a value and is always output as 21 for USER_DEFINED and the
userdeftype field would be required to supply the user defined name.

The following CEL backends (cel_odbc, cel_pgsql, cel_custom, cel_manager,
and cel_sqlite3_custom) can be independently configured to remove this
inconsistency.

* Allows cel_manager, cel_custom, and cel_sqlite3_custom to behave the
same way.

(closes issue ASTERISK-17189)
Reported by: Bryant Zimmerman

Review: https://reviewboard.asterisk.org/r/1669/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353648 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When ast_channel name was opaquified, the channel search functions did not
get converted correctly.  As a result ExtenSpy which uses a channel
iterator search by exten@context could never find anything.

* Updated the doxygen documentation for the search functions in channel.h.

Review: https://reviewboard.asterisk.org/r/1702/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353647 65c4cc65-6c06-0410-ace0-fbb531ad65f3

AST_AUDIOHOOK_TRIGGER_WRITE and AST_AUDIOHOOK_WANTS_DTMF were overlapping which
may have caused unintended side effects.  This patch moves
AST_AUDIOHOOK_TRIGGER_WRITE, and updates AST_AUDIOHOOK_TRIGGER_MODE to reflect
the original intention.

This will affect existing modules that use these flags, so be sure to recompile
as necessary.

(closes issue ASTERISK-19246)
Reported by: feyfre
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Clarified that using the VERBOSITY setting in etc_default_asterisk is the
same as using the -v command line switch, which causes Asterisk to launch
in console mode.

(closes issue ASTERISK-17030)
Reported by: Jonas
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The calendaring tech modules depend on res_calendar and initially
res_calendar just bumped the use count so that it couldn't be unloaded.
res_calendar can potentially create many threads and I've seen issues
where the Asterisk shutdown has failed where it looked like these
threads could be the culprit.

This patch adds unload support for res_calendar. Unloading res_calendar
will also unload the dependant tech modules as well.

(closes issue ASTERISK-16744)
Review: https://reviewboard.asterisk.org/r/1657/
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* Fix memory leak of vars in error paths for action_originate().

* Moved struct fast_originate_helper tech and data members to stringfields.

* Simplified ActionID header handling for fast_originate().

* Added doxygen note to ast_request() and ast_call() and the associated
channel callbacks that the data/addr parameters should be treated as const
char *.

Review: https://reviewboard.asterisk.org/r/1690/
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Asterisk's dnsmgr currently takes a pointer to an ast_sockaddr and updates it
anytime an address resolves to something different. There are a couple of
issues with this. First, the ast_sockaddr is usually the address of an
ast_sockaddr inside a refcounted struct and we never bump the refcount of those
structs when using dnsmgr. This makes it possible that a refresh could happen
after the destructor for that object is called (despite ast_dnsmgr_release
being called in that destructor). Second, the module using dnsmgr cannot be
aware of an address changing without polling for it in the code. If an action
needs to be taken on address update (like re-linking a SIP peer in the
peers_by_ip table), then polling for this change negates many of the benefits
of having dnsmgr in the first place.

This patch adds a function to the dnsmgr API that calls an update callback
instead of blindly updating the address itself. It also moves calls to
ast_dnsmgr_release outside of the destructor functions and into cleanup
functions that are called when we no longer need the objects and increments the
refcount of the objects using dnsmgr since those objects are stored on the
ast_dnsmgr_entry struct. A helper function for returning the proper default SIP
port (non-tls vs tls) is also added and used.

This patch also incorporates changes from a patch posted by Timo Teräs to
ASTERISK-19106 for related dnsmgr issues.

(closes issue ASTERISK-19106)

Review: https://reviewboard.asterisk.org/r/1691/
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https://origsvn.digium.com/svn/asterisk/branches/10

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  r353369 | alecdavis | 2012-01-31 11:42:28 +1300 (Tue, 31 Jan 2012) | 9 lines
  
  Merged revisions 353368 via svnmerge from 
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    r353368 | alecdavis | 2012-01-31 11:40:40 +1300 (Tue, 31 Jan 2012) | 2 lines
    
    prevent debug messsges displaying -ve Cseq numbers. Missed in R353320
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  r353321 | alecdavis | 2012-01-31 11:16:22 +1300 (Tue, 31 Jan 2012) | 25 lines
  
  Merged revisions 353320 via svnmerge from 
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    r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31 Jan 2012) | 18 lines
    
    RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer
    
    * fix: use %u instead of %d when dealing with CSeq numbers - to remove possibility of -ve numbers.
    
    * fix: change all uses of seqno and friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t.
    
    Summary of CSeq numbers.
    An initial CSeq number must be less than 2^31
    A CSeq number can increase in value up to 2^32-1
    An incrementing CSeq number must not wrap around to 0.
    
    Tested with Asterisk 1.8.8.2 with Grandstream phones.
     
    alecdavis (license 585)
    Tested by: alecdavis
     
    Review: https://reviewboard.asterisk.org/r/1699/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353319 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353317 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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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Fixes up softkey endcall. Previous code was a copy of onhook, now
allows for endcall softkey to be used while device is still onhook.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353224 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The previous change made the code look for emN and pciN in addition to what
it did originally, which was search for ethN.  However, it needed to be looking
for pciN#N, so that's what it does now.

This also moves the memset() to be before every ioctl().
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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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As of Fedora 15, ethN is not the name of ethernet interfaces.  The names
are emN or pciN.  Update some code that searched for interfaces named
ethN to look for the new names, as well.  For more information about why
this change was made, see this page:

    http://domsch.com/blog/?p=455
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Fix double format_cap iterator cleanup.
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I also went ahead and took a little time to make sure that the manager value
AMI_SUCCESS was used instead of just return 0 being thrown around everywhere since that's
how we handle this stuff these days.

(closes issue ASTERISK-19249)
Reporter: Jamuel Starkey
Patches:
	res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey (license 5766)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352981 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixes numerous reference leaks and missing ao2_iterator_destroy() calls as
a result.

Review: https://reviewboard.asterisk.org/r/1697/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352957 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds a CALENDAR_SUCCESS=1/0 variable that is set to show whether or
not CALENDAR_WRITE has passed. This patch also adds some debugging for caldav
PUT responses and no longer treats responses with no body as an error (as a PUT
gets a 201 Created with no body).

(closes issue ASTERISK-16903)
Reported by: Clod Patry
Tested by: Terry Wilson
Patches:
  	calendarstatus.diff uploaded by Clod Patry (License #5138), slightly modified by Terry Wilson

Review: https://reviewboard.asterisk.org/r/1692/
- This line, and those below, will be ignored--

M    res/res_calendar.c
M    res/res_calendar_exchange.c
M    res/res_calendar_caldav.c


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

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  r352863 | alecdavis | 2012-01-27 13:08:03 +1300 (Fri, 27 Jan 2012) | 19 lines
  
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    r352862 | alecdavis | 2012-01-27 13:05:30 +1300 (Fri, 27 Jan 2012) | 12 lines
    
    rfc4235 - Section 4.1: Versions MUST be representable using a non-negative 32 bit integer.
    
    If a BLF subscription exists for long enough, using %d may print negative version numbers.
    Unlikely, as 2^32 at 1 update per second is ~137 years, or half that before the versions number started going negative.
    
    Tested with Asterisk 1.8.8.2 with Grandstream phones.
     
    alecdavis (license 585)
    Tested by: alecdavis
     
    Review: https://reviewboard.asterisk.org/r/1694/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352864 65c4cc65-6c06-0410-ace0-fbb531ad65f3

sounds).

(Closes issue ASTERISK-19233)
Reported by: Matt Behrens
Patches:
        chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)
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For whatever reason, we don't have a single function for copying data like this
from SIP peers to the SIP pvt. This patch adds the copying of amaflags to the
sip_pvt, but it would probably be worth discussing this function along with
the others that essentially just copy some amount of data from a peer to a
private.

(Closes issue ASTERISK-19029)
Reported by: Matt Lehner
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https://origsvn.digium.com/svn/asterisk/branches/10

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  r352705 | alecdavis | 2012-01-26 19:33:11 +1300 (Thu, 26 Jan 2012) | 27 lines
  
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    r352704 | alecdavis | 2012-01-26 19:27:07 +1300 (Thu, 26 Jan 2012) | 20 lines
    
    Cleanup dialog-info+xml Notify dialog
    
    Make similar to other Notify messages.
    
    sample output:
    
    <?xml version="1.0"?>
    <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="715" state="full" entity="sip:8523@192.168.x.xx">
    <dialog id="8523">
    <state>terminated</state>
    </dialog>
    </dialog-info>
    
    Tested with Asterisk 1.8.8.2 with Grandstream phones.
     
    alecdavis (license 585)
    Tested by: alecdavis
     
    Review: https://reviewboard.asterisk.org/r/1693/
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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.


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........
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.
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* Fix authenticate MESSAGE losing custom headers added by the MESSAGE_DATA
function in the authorization attempt.

* Pass up better From header contents for SIP to use.  Now is in the
"display-name" <URI> format expected by MessageSend.  (Note that this is a
behavior change that could concievably affect some people.)

* Block user from adding standard headers that are added automatically.
(To, From,...)

* Allow the user to override the Content-Type header contents sent by
MessageSend.

* Decrement Max-Forwards header if the user transferred it from an
incoming message.

* Expand SIP short header names so the dialplan and other code only has to
deal with the full names.

* Documents what SIP expects in the MessageSend(from) parameter.

(closes issue ASTERISK-18992)
Reported by: Yuri

(closes issue ASTERISK-18917)
Reported by: Shaun Clark

Review: https://reviewboard.asterisk.org/r/1683/
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1) Be sure and free at unload the epa_backend we allocate at startup
2) Do the same sip_registry cleanup at unload we do at reload

Review: https://reviewboard.asterisk.org/r/1689/
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These files have no need to include "asterisk/version.h", and doing so forces
them to be rebuilt each time a Subversion checkout moves between 'modified'
and 'unmodified' states.
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There was faulty information in the sample config describing user as a synonym for friend
so it has been changed to better elaborate on the differences between the three entity
types.

(closes issue ASTERISK-15537)
Reported by: yarique
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352475 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue ASTERISK-16550)
reported by: Olle Johansson
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Now that we have the right license for the Russian 1.4.22 sounds as well as the sounds
for the Australian English 1.4.22 sounds, we can finally set the sounds to use 1.4.22!

(closes issue ASTERISK-18978)
Reported by: Cameron Twomey
Patches:
	confbridge.tar.001 uploaded by Cameron Twomey
    confbridge.tar.002 uploaded by Cameron Twomey
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Continue channel opaque-ification by wrapping all of the stringfields.
Eventually, we will restrict what can actually set these variables, but
the purpose for now is to hide the implementation and keep people from
adding code that directly accesses the channel structure. Semantic
changes will follow afterward.

Review: https://reviewboard.asterisk.org/r/1661/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fixed a potential memory leak when an existing datastore is manually
destroyed by inline code instead of calling ast_datastore_free().

(closes issue ASTERISK-17948)
Reported by: Archie Cobbs

Review: https://reviewboard.asterisk.org/r/1687/
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Merged revisions 352291 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 352292 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352293 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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Blocked revisions 352287

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Move RTP timeout check to before bridged channel check so it is actually executed.

(issue ASTERISK-19179)
Reported by: TSAREGORODTSEV Yury

(closes issue ASTERISK-14534)
Reported by: kriborgen
Patches:
	chan_sip.patch uploaded by kriborgen (license 6138)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352289 65c4cc65-6c06-0410-ace0-fbb531ad65f3