git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386462 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386461 65c4cc65-6c06-0410-ace0-fbb531ad65f3

* Make confbridge config parsing user profile, bridge profile, and menu
container hash/cmp functions correctly check the OBJ_POINTER, OBJ_KEY, and
OBJ_PARTIAL_KEY flags.

* Made confbridge load_module()/unload_module() free all resources on
failure conditions.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386375 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This crept in with the RESTful HTTP interface merge.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386352 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The two party bridging loops were changing the bridge peer pointers
without the channel locks held.  Thus when ast_channel_massquerade()
tested and used the pointer there is a small window of opportunity for the
pointers to become NULL even though the masquerade code has the channels
locked.

(closes issue ASTERISK-21356)
Reported by: William luke
Patches:
      jira_asterisk_21356_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: William luke
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386289 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Expand on a commit by OEJ to use the Coding-Guidelines

(issue ASTERISK-20259)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386266 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The API itself is documented using Swagger, a lightweight mechanism for
documenting RESTful API's using JSON. This allows us to use swagger-ui
to provide executable documentation for the API, generate client
bindings in different languages, and generate a lot of the boilerplate
code for implementing the RESTful bindings. The API docs live in the
rest-api/ directory.

The RESTful bindings are generated from the Swagger API docs using a set
of Mustache templates.  The code generator is written in Python, and
uses Pystache. Pystache has no dependencies, and be installed easily
using pip. Code generation code lives in rest-api-templates/.

The generated code reduces a lot of boilerplate when it comes to
handling HTTP requests. It also helps us have greater consistency in the
REST API.

(closes issue ASTERISK-20891)
Review: https://reviewboard.asterisk.org/r/2376/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386232 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Doxygen is only *ONE* comment that applies to the NEXT piece of code.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386211 65c4cc65-6c06-0410-ace0-fbb531ad65f3

sla.lock was already locked in the only place that sla_check_reload() was called.
Remove the redundant locking of sla.lock done in this function.  Less recursive
locking is A Good Thing.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386190 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There were several reports of deadlock when using
res_timing_pthread. Backtraces indicated that one thread was blocked
waiting for the write to the pipe to complete and this thread held
the container lock for the timers.  Therefore any thread that wanted
to create a new timer or read an existing timer would block waiting
for either the timer lock or the container lock and deadlock ensued.

This patch changes the way the pipe is used to eliminate this source
of deadlocks:

1) The pipe is placed in non-blocking mode so that it would never
block even if the following changes someone fail...

2) Instead of writing bytes into the pipe for each "tick" that's
fired the pipe now has two states--signaled and unsignaled. If
signaled, the pipe is hot and any pollers of the read side
filedescriptor will be woken up. If unsigned the pipe is idle. This
eliminates even the chance of filling up the pipe and reduces the
potential overhead of calling unnecessary writes.

3) Since we're tracking the signaled / unsignaled state, we can
eliminate the exta poll system call for every firing because we know
that there is data to be read.

(closes issue ASTERISK-21389)
Reported by: Matt Jordan
Tested by: Shaun Ruffell, Matt Jordan, Tony Lewis
patches:
  0001-res_timing_pthread-Reduce-probability-of-deadlocking.patch uploaded by sruffell (License 5417)

(closes issue ASTERISK-19754)
Reported by: Nikola Ciprich

(closes issue ASTERISK-20577)
Reported by: Kien Kennedy

(closes issue ASTERISK-17436)
Reported by: Henry Fernandes

(closes issue ASTERISK-17467)
Reported by: isrl

(closes issue ASTERISK-17458)
Reported by: isrl

Review: https://reviewboard.asterisk.org/r/2441/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386160 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This avoids some lock errors on the core set {debug,verbose} commands.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386054 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch adds the concept of ast_websocket_server to
res_http_websocket, allowing WebSocket connections on URL's more more
than /ws.

The existing funcitons for managing the WebSocket subprotocols on /ws
still work, so this patch should be completely backward compatible.

(closes issue ASTERISK-21279)
Review: https://reviewboard.asterisk.org/r/2453/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386020 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In messages.c, there are several places in the code where we create a
tmp_tech_holder and pass that into an ao2_find call. Unfortunately, we
weren't initializing the rwlock on the tmp_tech_holder, which the hash
function was locking. It's apparently harmless, but still not the best
code.

This patch extracts all that copy/pasted code into two functions,
msg_find_by_tech and msg_find_by_tech_name, which properly initialize
and destroy the rwlock on the tmp_tech_holder.

Review: https://reviewboard.asterisk.org/r/2454/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386019 65c4cc65-6c06-0410-ace0-fbb531ad65f3

res_xmpp and res_jabber need to search 'cachable' in the attrib section of the received IE, not data.

(issue ASTERISK-20175)
(closes issue ASTERISK-21429)
(closes issue ASTERISK-21069)
(closes issue ASTERISK-21164)

Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2452/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385939 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ast_enable_distributed_devstate is no longer applicable to how the
distributed device state system works and is no longer necessary.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385886 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Convert presence state events to Stasis-core messages and remove
redundant serializers where possible.

Review: https://reviewboard.asterisk.org/r/2410/


(closes issue ASTERISK-21102)
Patch-by: Kinsey Moore <kmoore@digium.com>


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385862 65c4cc65-6c06-0410-ace0-fbb531ad65f3

In the move from Asterisk's event system to Stasis, this makes
distributed device state aggregation always-on, removes unnecessary
task processors where possible, and collapses aggregate and
non-aggregate states into a single cache for ease of retrieval. This
also removes an intermediary step in device state aggregation.

Review: https://reviewboard.asterisk.org/r/2389/


(closes issue ASTERISK-21101)
Patch-by: Kinsey Moore <kmoore@digium.com>


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385860 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385835 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/2449/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385782 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385743 65c4cc65-6c06-0410-ace0-fbb531ad65f3

After some discussion on asterisk-dev, it was decided that the bulk of
the logic in app_stasis actually belongs in a resource module instead
of the application module.

This patch does that, leaves the app specific stuff in app_stasis, and
fixes up everything else to be consistent with that change.

 * Renamed test_app_stasis to test_res_stasis
 * Renamed app_stasis.h to stasis_app.h
   * This is still stasis application support, even though it's no
     longer in an app_ module. The name should never have been tied to
     the type of module, anyways.
 * Now that json isn't a resource module anymore, moved the
   ast_channel_snapshot_to_json function to main/stasis_channels.c,
   where it makes more sense.

Review: https://reviewboard.asterisk.org/r/2430/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385742 65c4cc65-6c06-0410-ace0-fbb531ad65f3

refactored to use these events rather than producing the events directly
in channel.c. Finally, the code was added to app_stasis to produce
DTMF events on the WebSocket.

The AMI events are completely backward compatible, including sending
events on transmitted DTMF, and sending DTMF start events.

The Stasis-HTTP events are somewhat simplified. Since DTMF start and
DTMF send events are generally less useful, Stasis-HTTP will only send
events on received DTMF end.

(closes issue ASTERISK-21282)
(closes issue ASTERISK-21359)
Review: https://reviewboard.asterisk.org/r/2439


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385734 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Normally I think keyword expansion is silly, but the one time it would have
been good, it didn't work because the property had quotes in it. This patch
fixes obviously busted svn:keywords properties.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385718 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch calculates the timestamp for outbound RTP when we don't have timing
information. This uses the same approach in res_rtp_asterisk. Thanks to both
Pietro and Tzafrir for providing patches.

(closes issue ASTERISK-19883)
Reported by: Giacomo Trovato
Tested by: Pietro Bertera, Tzafrir Cohen
patches:
  rtp-timestamp-1.8.patch uploaded by tzafrir (License 5035)
  rtp-timestamp.patch uploaded by pbertera (License 5943)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385638 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior to this patch, a read error in snd_pcm_readi would still be treated as a
nominal result when constructing a voice frame from the expected data. Since
the value returned is negative, as opposed to the number of samples read,
this could result in a crash. With this patch, we now return a null frame
when a read error is detected.

Note that the patch on ASTERISK-21329 was modified slightly for this commit,
in that we bail immediately on detecting the read error, rather than bypassing
the construction of the voice frame.

(closes issue ASTERISK-21329)
Reported by: Keiichiro Kawasaki
patches:
  chan_alsa.diff uploaded by kawasaki (License 6489)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385635 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When app_queue is unloaded, some manager commands are not being unregistered
which result in a segfault.  This patch corrects this.

(closes issue ASTERISK-21397)
Reported by: Peter Katzmann, Corey Farrell
Tested by: Corey Farrell
Patches:
    asterisk-21397-missing-unreg-manager-cmd_1.8.diff
                                                 Michael L. Young (license 5026)
    asterisk-21397-missing-unreg-manager-cmd_11.diff
                                                 Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2444/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385595 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Ensure that trans_size is correct to prevent uninitialized entries from
preventing reload.

(closes issue ASTERISK-21401)
Reported by: Corey Farrell
Tested by: Corey Farrell
Patches:
    codec_resample-unload.patch uploaded by Corey Farrell
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385585 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The original report was that app_voicemail would crash.  This was caused by
ast_config_load() returning CONFIG_STATUS_FILEINVALID but no checks being
performed for that return status.  After adding the initial patch to fix this
issue, Jaco Kroon (jkroon) added some fixes to memory leaks he had discovered.

During review, Walter Doekes (wdoekes) suggested adding a helper function in
order to determine if we had a valid configuration or not.

This patch does the following:

* Creates a helper function to check if the configuration is valid

* Adds calls to the new helper function where appropiate

* Fixes memory leaks where the code returned without running
  ast_config_destroy() on the configuration that was loaded

(closes issue ASTERISK-21302)
Reported by: Jaco Kroon
Tested by: Jaco Kroon, Michael L. Young
Patches:
    asterisk-11.3.0-app_voicemail-ast_config-fixes.patch
                                                       Jaco Kroon (license 5671)
    asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff
                                                 Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2443/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385573 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385548 65c4cc65-6c06-0410-ace0-fbb531ad65f3

These functions are already used in one branch (jrose's parking branch)
and will soon be used in other branches as well.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385522 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When we reload Asterisk or chan_sip, the flags force_rport and comedia that are
turned on and off when using the auto_force_rport and auto_comedia nat settings
go back to the default setting off.  These flags are turned on when needed or
off when not needed at the time that a peer registers, re-registers or initiates
a call.  This would apply even when only the default global setting
"nat=auto_force_rport" is being used, which in this case would only affect the
force_rport flag.

Everything is good except for the following:  The nat setting is set to
auto_force_rport and auto_comedia.  We reload Asterisk and the peer's
registration has not expired.  We load in the settings for the peer which turns
force_rport and comedia back to off.  Since the peer has not re-registered or
placed a call yet, those flags remain off.  We then initiate a call to the peer
from the PBX.  The force_rport and comedia flags stay off.  If NAT is involved,
we end up with one-way audio since we never checked to see if the peer is behind
NAT or not.

This patch does the following:

* Moves the checking of whether a peer is behind NAT into its own function

* Create a function to set the peer's NAT flags if they are using the auto_* NAT
  settings

* Adds calls in sip_request_call() to these new functions in order to setup the
  dialog according to the peer's settings

(closes issue ASTERISK-21374)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
    asterisk-21374-auto-nat-outgoing-fix_v2.diff Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2421/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385474 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Ensure iax2_process_thread is signalled when a deferred frame is queued to it.

(closes issue ASTERISK-18827)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2426/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385431 65c4cc65-6c06-0410-ace0-fbb531ad65f3

On startup, it's possible for a frame to arrive before the processing threads were ready.

In iax2_process_thread() the first pass through falls into ast_cond_wait, should a frame arrive
before we are at ast_cond_wait, the signal will be ignored.
The result iax2_process_thread stays at ast_cond_wait forever, with deferred frames being queued.  

Fix: When creating initial idle iax2_process_threads, wait for init_cond to be signalled
after each thread is started.
 
(issue ASTERISK-18827)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)

Review https://reviewboard.asterisk.org/r/2427/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385406 65c4cc65-6c06-0410-ace0-fbb531ad65f3

........
Add dependency on libuuid, for res_rtp_asterisk

pjproject is what actually requires libuuid.

(closes issue ASTERISK-21125)
reported by Private Name

(Ed. note: Really?  Private Name?  I am rolling my eyes so hard right now.)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385357 65c4cc65-6c06-0410-ace0-fbb531ad65f3

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385314 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385278 65c4cc65-6c06-0410-ace0-fbb531ad65f3

protected by the features_lock.

* Made all calls to ast_find_call_feature() have the features_lock held.

* Fixed set_config_flags() to actually use find_group() to look for
feature groups in DYNAMIC_FEATURES.  The code originally assumed all
feature groups were listed in DYNAMIC_FEATURES.

* Make everyone use ast_rdlock_call_features(),
ast_unlock_call_features(), and new ast_wrlock_call_features() instead of
directly calling the rwlock API on features_lock.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385277 65c4cc65-6c06-0410-ace0-fbb531ad65f3

For the events that have been ported to Stasis, this was broken in
r384910, when a couple of lines of code was lost in a merge.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385236 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When MALLOC_DEBUG is enabled with res_config_ldap, issues (munmap_chunk:
invalid pointer errors) can occur as the memory is being allocated with
Asterisk's wrappers around malloc/calloc/free/strdup, as opposed to the
LDAP library's wrappers.

This patch uses the LDAP library's wrappers where appropriate, so that
compiling with MALLOC_DEBUG doesn't cause more problems than it solves.

Note that the patch listed below was modified slightly for this commit
to account for some additional memory allocation/deallocations.

(closes issue ASTERISK-17386)
Reported by: John Covert
Tested by: Andrew Latham
patches:
  issue18789-1.8-r316873.patch uploaded by seanbright (License 5060)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385202 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When a BYE request is processed in chan_sip, the current SIP dialog is detached
from its associated Asterisk channel structure. The tech_pvt pointer in the
channel object is set to NULL, and the dialog persists for an RFC mandated
period of time to handle re-transmits.

While this process occurs, the channel is locked (which is good).
Unfortunately, operations that are initiated externally have no way of knowing
that the channel they've just obtained (which is still valid) and that they are
attempting to lock is about to have its tech_pvt pointer removed. By the time
they obtain the channel lock and call the channel technology callback, the
tech_pvt is NULL.

This patch adds a few checks to some channel callbacks that make sure the
tech_pvt isn't NULL before using it. Prime offenders were the DTMF digit
callbacks, which would crash if AMI initiated a DTMF on the channel at the
same time as a BYE was received from the UA. This patch also adds checks on
sip_transfer (as AMI can also cause a callback into this function), as well
as sip_indicate (as lots of things can queue an indication onto a channel).

Review: https://reviewboard.asterisk.org/r/2434/

(closes issue ASTERISK-20225)
Reported by: Jeff Hoppe
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385174 65c4cc65-6c06-0410-ace0-fbb531ad65f3