* Fix threadpool_alloc() prototype.

* Add missing off-nominal NULL check of pool in threadpool_alloc().

* searializer_create() does not need to create the object with a lock as
the lock is not used.
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Due to the architecture of how media streams are handled each individual
handler adds connection details (IP address) for it. The first media stream
is then used as the top level SDP connection line. In practice each
line ends up being the same so to reduce the SDP size stream-level connection
information is also added to the SDP if it differs from the top level SDP
connection line.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424077 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There are certain situations which no checks existed for which need to prevent
session refreshes. This includes sending a session refresh with SDP before SDP
negotiation has completed and sending a session refresh before the dialog itself
has been established. Checks for these have been added.

Additionally COLP related UPDATEs were including SDP when it is not needed.

Review: https://reviewboard.asterisk.org/r/4008/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424039 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424038 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-24357 #close
Reported by: Jeremy Laine
Patches:
      res_fax_bounds.patch (license #6561) patch uploaded by Jeremy Laine
	  Modified patch to not use magic numbers.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424016 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This gets rid of most old libc free/malloc/realloc and replaces them
with ast_free and friends. When compiling with MALLOC_DEBUG you'll
notice it when you're mistakenly using one of the libc variants. For
the legacy cases you can define WRAP_LIBC_MALLOC before including
asterisk.h.

Even better would be if the errors were also enabled when compiling
without MALLOC_DEBUG, but that's a slightly more invasive header
file change.

Those compiling addons/format_mp3 will need to rerun
./contrib/scripts/get_mp3_source.sh.

ASTERISK-24348 #related
Review: https://reviewboard.asterisk.org/r/4015/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423978 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ASTERISK-23768 #close
Reported by: Jeremy Lainé
Patches:
  escape_manpage_hyphen.patch uploaded by Jeremy Lainé (License #6561)
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* Made memset the std struct in ast_sip_push_task_synchronous() because if
DEBUG_THREADS is enabled then uninitialized lock tracking data is used.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423896 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The new option 'preferchannelclass' is added to musiconhold.conf. If yes
(the default) the CHANNEL(musicclass) is preferred when choosing the
hold music. If it is no, the class suggested by the application that
calls the MoH (e.g. the Queue() app) gets preferred (new behaviour).

This way you set a different hold-music from the Queue-music by setting
both the CHANNEL(musicclass) and the queue-context musicclass.

ASTERISK-24276 #close
Reported by: Kristian Høgh
Patches:
  app_override_channel_moh.patch uploaded by Kristian Høgh (License #6639)

Review: https://reviewboard.asterisk.org/r/4010/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423893 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The crash on the issues is a result of an invalid transport configuration
change when asterisk is restarted.  The attempt to send the qualify
request fails and we cleaned up.  However, the callback is also called
which results in a double unref of the objects involved.

* Put a wrapper around pjsip_endpt_send_request() to detect when the
passed in callback is called because of an error so callers can know to
not cleanup.

* Made send_request_cb() able to handle repeated challenges (Up to 10).

* Fix periodic endpoint qualify OPTIONS sched deletion race by avoiding
it.  The sched entry will no longer self stop and must be externally
stopped.

* Added REF_DEBUG description tags to struct sched_data in
pjsip_options.c.

* Fix some off-nominal ref leaks in schedule_qualify(),
qualify_and_schedule().

* Reordered pjsip_options.c module start/stop code to cleanup better on
error.

ASTERISK-24295 #close
Reported by: Rogger Padilla

Review: https://reviewboard.asterisk.org/r/3954/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423868 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Make sure outbound proxy refs are always unreffed on dialog destruction.

Review: https://reviewboard.asterisk.org/r/4016/
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Prior to this commit, CDR and CEL tests were expected to trigger
FRACKs (i.e. assertions) due to the fact that the channels they
create have no formats on them. Some code was independently added
recently that attempts to prevent FRACKs from occurring by failing
early when attempting to set up translation paths if one or both
channels support no formats. Unfortunately, this attempt to be helpful
made the CDR and CEL tests go from simply FRACKing to outright
failing and in some cases, failing so badly as to crash Asterisk.

This commit seeks to correct past mistakes by adding the ulaw format
to channels created by the CDR and CEL unit tests. This makes setting
up translation paths succeed, eliminates previously-seen FRACKs, and
ultimately causes the unit tests to succeed again.

Review: https://reviewboard.asterisk.org/r/4014
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423784 65c4cc65-6c06-0410-ace0-fbb531ad65f3

INVITE arrives to asterisk, asterisk responds Busy(). If the INVITE is
retransmitted, asterisk would generate a 503 in addition to the 486.

Thanks Torrey Searle for providing a working regression test.

ASTERISK-24335 #close

Review: https://reviewboard.asterisk.org/r/4003/
Patches:
  retrans_486_invite.patch uploaded by Torrey Searle (License #5334)
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r421600 conflicted with r155763.

ASTERISK-24348 #close
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In r423414 (13) / r423415 (trunk), an API call that determines if a format
capability structure is empty was added. This returns true if the format
capability structure is completely empty or "none". A check for this was added
in channel.c's set_format call. Unfortunately, when this check was true, it
returned from the function while still holding the channel lock. This caused
the CDR unit tests - which have a tendency to create channels with no formats -
to deadlock. Whoops.

This patch unlocks the channel on the off-nominal path.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423642 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior to the release of Swagger 1.2, the attribute 'extends' was being
promoted as a possible way to show that a particular object extends an existing
object. Instead, the Swagger specification went with the 'subTypes' attribute
in the base object. This patch removes the unsupported attribute; the object
that the offending objects proposed to extend already lists them in its
'subTypes' attribute.

ASTERISK-24300 #close
Reported by: Bradley Watkins
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423622 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The resources.json file that defines the resource JSON files used with ARI
references a basePath of 'http://localhost:8088/ari'. This does not match what
is defined in the resource files themselves, 'http://localhost:8088/stasis'.
The correct base path is the one that includes 'ari' in the URL; this patch
updates the various resource JSON files to have the correct basePath.

ASTERISK-24339 #close
Reported by: Bradley Watkins
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When unloading the module did not unregister the CLI commands causing a crash upon
load when they were registered again.

When reloading the module the return value from the config options framework was not
checked to determine if an error occurred or not. This caused a message to be output
saying the module did not exist when reloading if no changes were present.

AST-1433 #close
AST-1434 #close
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423581 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Outgoing PJSIP calls can result in non-negotiated formats listed in the
channel's native formats if video formats are listed in the endpoint's
configuration.  The resulting call could then use a non-negotiated format
resulting in one way audio.

* Simplified the update of session->req_caps in set_caps().  Why do
something in five steps when only one is needed?

AFS-162 #close

Review: https://reviewboard.asterisk.org/r/4000/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423563 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Masquerades into channels that are in the dialing state don't end their dial
and this goes against the model for things like CDRs and generating Dial end
manager actions and such.

ASTERISK-24237 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/3990/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423546 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Caused by format changes in Asterisk 13

ASTERISK-24265 #close
Reported by: Dafi Ni
Review: https://reviewboard.asterisk.org/r/3999/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423526 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This change gives framehooks a reverse-direction masquerade callback in
addition to chan_fixup_cb similar to the callback added to datastores
to handle the same situation. The new callback provides the same
parameters as the fixup callback, but is called on the new channel's
framehooks before moving framehooks from the old channel to the new
channel. This gives the framehooks an oppurtunity to decide whether
they should remain on the new channel or be removed.

This new callback is used to prevent the PJSIP T.38 framehook from
remaining on a masqueraded channel if the new channel is not also a
PJSIP channel. This was causing a crash when a local channel was
masqueraded into a PJSIP channel and the framehook was executed on the
local channel since the channel's tech private data was not structured
as expected.

Review: https://reviewboard.asterisk.org/r/4001/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423505 65c4cc65-6c06-0410-ace0-fbb531ad65f3

An empty password is valid for username/password authentication so we should
allow password to be empty/not supplied.

Review: https://reviewboard.asterisk.org/r/3988
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423483 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This function acts like strsep with three exceptions...
* The separator is a single character instead of a string.
* Separators inside quotes are treated literally instead of like separators.
* You can elect to have leading and trailing whitespace and quotes
stripped from the result and have '\' sequences unescaped.

Like strsep, ast_strsep maintains no internal state and you can call it
recursively using different separators on the same storage.

Also like strsep, for consistent results, consecutive separators are not
collapsed so you may get an empty string as a valid result.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3989/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423480 65c4cc65-6c06-0410-ace0-fbb531ad65f3

These are needed for a set of batched notification RLS tests that are
about to be committed to the testsuite.

Review: https://reviewboard.asterisk.org/r/3967
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423463 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Also fixes comma separates match lists

ASTERISK-24290 #close
Reported by: Ray Crumrine
Review: https://reviewboard.asterisk.org/r/3995/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423442 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Clarified some read/write format comments.

* Fixed a doxygen tag typo.
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* Make astob2 REF_DEBUG output an invalid object line when an invalid ao2
object ref/unref is attempted.  This is similar to the
constructor/destructor lines.

* Fixed refcounter.py to handle skewed objects that have
constructor/destructor states.

* Made refcounter.py highlight the invalid ao2 object refs by putting them
in their own section of the processed output file.

* Made refcounter.py highlight unreffing an object by more than one that
results in a negative ref count and the object being destroyed.  The
abnormally destroyed object is reported in the invalid and finalized
object sections of the output.

Review: https://reviewboard.asterisk.org/r/3971/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423422 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Empty here means that there are no formats in the format_cap structure
or the only format in it is the "none" format.

I've added calls to check the emptiness of a format_cap in a few places
in order to short-circuit operations that would otherwise be pointless
as well as to prevent some assertions from being triggered in cases
where channels with no formats are used.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423415 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If faxing fails at a very early stage, then it is possible for
us to pass a NULL t30 state pointer to spandsp, which spandsp
is none too pleased with.

This patch ensures that we pass the correct pointer to spandsp
in the situation where we have not yet set our local t30 state
pointer.

ASTERISK-24301 #close
Reported by Matt Jordan
Patches:
	ASTERISK-24301-fax.diff Uploaded by Mark Michelson (License #5049)
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res_pjsip_pubsub has two separate checks that it makes when a SUBSCRIBE
arrives.
* It checks that there is a subscription handler for the Event
* It checks that there are body generators for the types in the Accept header

The problem is, there's nothing that ensures that these two things will
actually mesh with each other. For instance, Asterisk will accept a subscription
to MWI that accepts pidf+xml bodies. That doesn't make sense.

With this commit, we add some type information to the mix. Subscription
handlers state they generate data of type X, and body generators state
that they consume data of type X. This way, Asterisk doesn't end up in
some hilariously mismatched situation like the one in the previous paragraph.

ASTERISK-24136 #close
Reported by Mark Michelson

Review: https://reviewboard.asterisk.org/r/3877
Review: https://reviewboard.asterisk.org/r/3878
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423350 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When no transport is associated to an endpoint, the AMI output for
PJSIPShowEndpoint indicates an error instead of silently ignoring the
missing transport.

This patch causes the error to appear only if a transport was specified
on the endpoint and the transport doesn't exist.  It also fixes an issue
with counting the objects that were actually found.

ASTERISK-24161 #close
ASTERISK-24331 #close
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3998/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423285 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch changes the install to only install the hook script if
DAHDI is enabled. It also adds the script to the uninstall task, and
moves the DAHDI_UDEV_HOOK_DIR variable so that it's not between the
_MAKEOPTS variables and their comment.

This allows installs which specify a --prefix to work normally, as
long as they don't enable DAHDI.

Review: https://reviewboard.asterisk.org/r/3972/
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If you call ast_category_insert with a match category that doesn't exist, the
list traverse runs out of 'next' categories and you get a SEGV.  This patch
adds check for the end-of-list condition and changes the signature to return
an int for success/failure indication instead of a void.

The only consumer of this function is manager and it was also changed to use
the return value.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3993/
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This patch was supposed to go into a team branch, but I was a bit fast on the
gun before 'svn switch' had apparently moved the target branch over.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423238 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423237 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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  r423209 | file | 2014-09-16 17:35:34 -0300 (Tue, 16 Sep 2014) | 8 lines
  
  res_rtp_asterisk: Fix building when pjproject is not used.
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  Merged revisions 423207 from http://svn.asterisk.org/svn/asterisk/branches/11
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  Merged revisions 423208 from http://svn.asterisk.org/svn/asterisk/branches/12
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  r423212 | file | 2014-09-16 18:03:59 -0300 (Tue, 16 Sep 2014) | 10 lines
  
  res_rtp_asterisk: Fix 100% CPU usage due to timer heap thread spinning.
  
  Side note: I need a vacation.
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  Merged revisions 423210 from http://svn.asterisk.org/svn/asterisk/branches/11
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  Merged revisions 423211 from http://svn.asterisk.org/svn/asterisk/branches/12
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Merged revisions 423209,423212 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423213 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Changes made during format improvements resulted in the
recording to voicemail option 'm' of the MixMonitor app
writing a zero length duration in the msgXXXX.txt file.

This change introduces a new function ast_ratestream(),
which provides the sample rate of the format associated
with the stream, and updates the app_voicemail function
for ast_app_copy_recording_to_vm to calculate the right
duration.

Review: https://reviewboard.asterisk.org/r/3996/
ASTERISK-24328 #close
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Merged revisions 423192 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423193 65c4cc65-6c06-0410-ace0-fbb531ad65f3