git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381628 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Previously, presencestate information was sent whenever the state was not
NOT_SET. When r381594 actually returned INVALID presence state in all the
places it was supposed to, it caused chan_sip to start adding presence
state information to NOTIFY requests that it previously would not have
added. chan_sip shouldn't be adding presence state information when the
provider is in an invalid state; users can't set the state to invalid and
an invalid state always implies that the provider is in an error condition.

(issue AST-1084)
........

Merged revisions 381613 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381615 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/2329/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381614 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch fixes a crash in Asterisk that could be caused by using the
PresenceState AMI action while providing an invalid provider. This patch
also adds some additional warnings when a user attempts to provide the
PresenceState action with invalid data, and removes some NOTICE statements
that were still lurking in the code from testing.

(closes issue AST-1084)
Reported by: John Bigelow
Tested by: John Bigelow
........

Merged revisions 381594 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381595 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Reference counting for the channel and its tech_pvt got messed up at
some point between 1.8 and 11. The result was that if a BYE for a dialog
that had been replaced (via an INVITE with Replaces) was received, Asterisk
would crash due to trying to access data on a channel that was no longer there.

The fix I introduced is to remove code that both unrefs the sip_pvt and sets
the channel's tech_pvt to NULL when an INVITE with Replaces is handled. This
way when a BYE is received, the tech_pvt will be non-NULL and so the BYE can
be processed and not cause a crash.

(closes issue ASTERISK-20929)
reported by Kristopher Lalletti
patches:
	ASTERISK-20929.patch uploaded by Mark Michelson (License #5049)
........

Merged revisions 381566 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381568 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does two things:
 1. It disables (temporarily) strict XML documentation checking for module
    configurations. We should re-enable it before making any release from
    trunk.
 2. Pass the module flag AST_MODULE through sorcery. This means several of the
    API calls are now macros and will do this automatically for you. The config
    framework needs the module that objects are registering to so it can
    properly construct the documentation. (This was already a required field,
    but sorcery was getting by without it)




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381567 65c4cc65-6c06-0410-ace0-fbb531ad65f3

While autoservice is running and servicing a channel the callid is being stored
and removed in the thread's local storage for each iteration of the thread loop.
If debug was set to a sufficient level the log file would be spammed with callid
thread local storage debug messages.

Added a new function that checks to see if the callid to be stored is different
than what is already contained (if anything).  If it is different then
store/replace and log, otherwise just leave as is.  Also made it so all logging
of debug messages pertaining to the callid thread storage outputs only when
TEST_FRAMEWORK is defined.

(issue ASTERISK-21014)
(closes issue ASTERISK-21014)
Report by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2324/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381557 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Some bad copy/pasting resulted in using the audio crypto attribute for both
text and video RTP. Also the audio crypto isn't set until after these, so it
was really just bad all around.

(closes ASTERISK-20905)
Reported by: Kristopher Lalletti
patches:
	rtp_crypto_video_text.diff uploaded by Jonathan Rose (license 6182)
........

Merged revisions 381553 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381556 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381543 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch allows a module to define its configuration in XML in source, such
that it can be parsed by the XML documentation engine. Documentation is
generated in a two-pass approach:

1. The documentation is first generated from the XML pulled from the source
2. The documentation is then enhanced by the registration of configuration
   options that use the configuration framework

This patch include configuration documentation for the following modules:
 * chan_motif
 * res_xmpp
 * app_confbridge
 * app_skel
 * udptl

Two new CLI commands have been added:
 * config show help - show configuration help by module, category, and item
 * xmldoc dump - dump the in-memory representation of the XML documentation to
   a new XML file.

Review: https://reviewboard.asterisk.org/r/2278
Review: https://reviewboard.asterisk.org/r/2058

patches:
  on review 2058 uploaded by twilson



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381527 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381471 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Don't know why it seemed to work during testing, but it really is needed 
for protocol v17 (and probably above).


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381470 65c4cc65-6c06-0410-ace0-fbb531ad65f3

It doesn't hurt to check AST_SOFTHANGUP_UNBRIDGE either, but it should not
be set outside of a bridge.

(issue ASTERISK-20492)
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Merged revisions 381466 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 381467 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381469 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix chan_skinny so that it respects callerID presentation of inbound calls to 
device and a couple of other minor fixes: 145 packet (add OCTAL_FROM amd callerid),
and dont send dialednumber message if protocol >= 17. 

(closes issue ASTERISK-21066)
Reported by: snuffy
Tested by: snuffy, myself
Patches: 
    skinny-respect-clid-restrictions-v2.diff uploaded by snuffy (license 5024)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381465 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The core module related to coloring terminal output was old and needed
some love.  The main thing here was an attempt to get rid of the
obscene number of stack-local buffers that were allocated for no other
reason than to colorize some output.  Instead, this uses a simple trick
to allocate several buffers within threadlocal storage, then
automatically rotates between them, so that you can make multiple calls
to the colorization routine within one function and not need to
allocate multiple buffers.

Review: https://reviewboard.asterisk.org/r/2241/
Patches:
    bug.patch uploaded by Tilghman Lesher


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381448 65c4cc65-6c06-0410-ace0-fbb531ad65f3

While adding red-black tree containers to astobj2 in r376575, Richard pointed
out the way chan_iax2 finds unused call numbers will prevent ao2_container
integrity checks at runtime.

This patch removes the ao2_container and instead uses fixed sized arrays and a
modified Fisher-Yates-Durstenfeld shuffle to maintain the call number list.

While the locking semantics are similar to the ao2_container implementation,
this implementation should be faster and more memory efficient.

Review: https://reviewboard.asterisk.org/r/2288/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381427 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381398 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The function call ast_db_deltree returns the number of row deleted, or a
negative number if it failed. DBdeltree was treating any non-zero return
as an error, causing a spurious verbose error message to be displayed.

This patch handles the return code of ast_db_deltree correctly.

(closes issue ASTERISK-21070)
Reported by: ianc
patches:
  dbdeltree.diff uploaded by ianc (License #5955)
........

Merged revisions 381364 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 381365 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381366 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds the ability to create a serializer from a thread pool. A
serializer is a ast_taskprocessor with the same contract as a default
taskprocessor (tasks execute serially) except instead of executing out
of a dedicated thread, execution occurs in a thread from a
ast_threadpool. Think of it as a lightweight thread.

While it guarantees that each task will complete before executing the
next, there is no guarantee as to which thread from the pool individual
tasks will execute. This normally only matters if your code relys on
thread specific information, such as thread locals.

This patch also fixes a bug in how the 'was_empty' parameter is computed
for the push callback, and gets rid of the unused 'shutting_down' field.

Review: https://reviewboard.asterisk.org/r/2323/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381326 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The RTP engine will no longer allow for local and remote native RTP bridges
if packetization of streams differs. Allowing native bridging in this scenario
has been known to cause FAX failures.

(closes ASTERISK-20650)
Reported by: Maciej Krajewski
Patches:
	ASTERISK-20659.patch uploaded by Mark Michelson (License #5049)

Review: https://reviewboard.asterisk.org/r/2319
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Merged revisions 381281 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 381306 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381307 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When sip_ref_peer and sip_unref_peer were exported to be usable in
channels/sip/security_events.c, modifications to those functions when
building under REF_DEBUG were not taken into account. This change
moves the necessary defines into sip.h to make them accessible to
other parts of chan_sip that need them.
........

Merged revisions 381282 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381285 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Currently, ConfBridge does not send manager events for ConfbridgeMute,
ConfbridgeUnmute, ConfbridgeStartRecord and ConfbridgeStopRecord.  This patch
adds these events to the manager.

The reporter's patch moves some other events up to the beginning of the file.

The patch being committed is based on the patch contributed from the reporter of
this issue.  I have made a lot of modifications to the patch in order for it to
fit in better with what we currently are doing in the code when it comes to
manager events.  I also made a few changes to the <see-also> elements on some of
the events.

(closes issue ASTERISK-20827)
Reported by: Clint Davis
Tested by: Clint Davis, Michael L. Young
Patches:
    20827.diff uploaded by Clint Davis (license 6453)
    asterisk-20827-confbridge-events.diff uploaded by 
                                                 Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2309/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381256 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If say.conf did not exists prior to originally loading module app_playback it
would not load on subsequent reloads of the module once it had been created.
This occurred because upon reload of the app_playback module it would only
load a new configuration if an old one had previously existed.  This fix simply
removed the association between checking if an old configuration existed and
the loading of the new one.

(closes issue ASTERISK-20800)
Reported by: pgoergler
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Merged revisions 381216 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 381217 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381219 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When the red/black tree work was committed, there was an extra ", " in
the REF_DEBUG definition of ao2_container_alloc_rbtree.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381218 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Made input checking more consistent with other Asterisk code
* Added validation to ast_json_dump_new_file
* Fixed tests for ownereship semantics

(issue ASTERISK-20887)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381214 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add extra string to transmit_callinfo_var, Only set string2 to tonum for outgoing calls
and changes to send_callinfo and push_callinfo to not set callid name to last number.

(closes issue ASTERISK-21063)
Reported by: wedhorn
Tested by: snuffy, myself
Patches: 
    skinny-callinfoupdate03.diff uploaded by wedhorn (license 5019)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381195 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Caching a struct ast_app pointer is not a good idea because someone could
unload the application.  After the applicaiton unload the cached ast_app
pointer is no longer valid.  Only pbx.c can cache the pointer because it
knows when the application is unloaded and removes the pointer.

* Fixed one-touch Monitor and MixMonitor to not cache the ast_app pointer
and not use the silly monitor_ok/mixmonitor_ok/stopmixmonitor_ok flags.

* Extracted bridge_check_monitor() from ast_bridge_call() and use propper
locking.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381177 65c4cc65-6c06-0410-ace0-fbb531ad65f3

An error existed in res_xmpp where it would attempt to delete attributes from
a node that itself was also deleted. Per the iksemel documentation, attributes
added using iks_insert are copied to the parent node's stack, and will be
reclaimed when that node is itself destroyed.

(closes issue ASTERISK-20982)
Reported by: marcelloceschia
patches:
  delete-node-fix.diff uploaded by marcelloceschia (License 6036)
........

Merged revisions 381159 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381160 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This commit adds native implementation support for copying and diffing objects,
as well as the ability to load or reload on a per-object type level.

Review: https://reviewboard.asterisk.org/r/2320/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381134 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Taking advantage of the sorted order of the registered functions container
requires that they are actually inserted in the expected sort order.

* Insert the registered functions into the container in case sensitive
position.  As a result, only the complete_functions() routine needs to
search the entire container because it does a case insensitive search for
convenience.

Caught by the unit tests.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381118 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Fixed "core show function" tab completion and token count checking.

* Refactored function and application container handling code to reduce
redundancy.

* Made __ast_pbx_run() return using the defines the caller should expect.
Doesn't change the returned values.  Just made use the defines.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381102 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381086 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Unloading ConfBridge caused the next AMI action received to crash
Asterisk.

* Add the missing unregister of AMI action ConfbridgeSetSingleVideoSrc
when ConfBridge is unloaded.

(closes issue ASTERISK-20994)
Reported by: Jeremy Kister
Patches:
      jira_asterisk_20994_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: Rusty Newton, Jeremy Kister
........

Merged revisions 381067 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381069 65c4cc65-6c06-0410-ace0-fbb531ad65f3

These two variables were previously not being set when comebacktoorigin=yes
and the example configs seemed to imply that they should be. Since there
is no harm in this and since calls that are sent back to origin are capable
of continuing in the dialplan, this seemed like a no-brainer. Also it
supports some bridging tests I've been working on.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381068 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Since res_sorcery_config used a static name of "res_sorcery_config" to inform the configuration
file API that it asked for the configuration file it was possible during a reload for some sorcery
object types not to receive the new configuration file.

This change introduces a UUID on a per-sorcery config instance basis so that the unchanged state
is kept on an instance basis and not for the res_sorcery_config module as a whole.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381037 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add struct-level comparison and copying of string fields to reduce the
complexity of whole-struct comparison and copying when using string
fields. The new macros do not take into account non-stringfield data.

Review: https://reviewboard.asterisk.org/r/2308/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381017 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When I added my extensive suite of session timer unit tests, apparently one of
them was failing and I never noticed. If neither Min-SE nor Session-Expires is
set in the header, it was responding with a Session-Expires of the global
maxmimum instead of the configured max for the endpoint.

(issue ASTERISK-20787)
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Merged revisions 380973 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380977 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Patch ensures that d->activeline and l->activesub are moved over to the
new device and line so that on callend the appropriate subs can be found
to complete hangup before device resets.

(closes issue ASTERISK-16610)
Reported by: wedhorn
Tested by: snuffy, myself
Patches: 
    skinny-reloadactive01.diff uploaded by wedhorn (license 5019)
........

Merged revisions 380942 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380943 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Make skinny reset vmexten and immeddial to '\0' on reload to ensure that
it is set to '\0' if the appropriate item is removed/commented in 
skinny.conf. Also small fix re immeddial char in skinny.conf and add
immedial setting to skinny show settings.

(closes issue ASTERISK-21037)
Reported by: snuffy
Tested by: snuffy, myself
Patches: 
    immed_dial_fix.diff uploaded by snuffy (license 5024)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380925 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The Page and ConfBridge custom announcement did not play when users
entered the conference.

* Fix the CONFBRIDGE(user,announcement) file not getting played.  The code
to do this got removed accidentally when the ConfBridge code was
restructured to be more state machine like.

* Fixed play_prompt_to_user() doxygen comments.

* Fixed the Page A(x) and n options for the caller.  The caller never
played the announcement file and totally ignored the n option.  The code
to do this was lost when the application was converted to use ConfBridge.

* Factored out setup_profile_bridge(), setup_profile_paged(), and
setup_profile_caller() routines to setup ConfBridge profiles.  Made each
profile setup routine use the default template if one has not already been
setup by dialplan.

(closes issue ASTERISK-20990)
Reported by: Jeremy Kister
Tested by: rmudgett
........

Merged revisions 380894 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380896 65c4cc65-6c06-0410-ace0-fbb531ad65f3