Review: https://reviewboard.asterisk.org/r/2449/
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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@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

Because "struct feature_ds *feature_ds" is not a good thing.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385142 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The hash and compare functions for the control container was reusing
the wrong ones, causing some problems. I fixed it, but in the wrong
branch. Oh well, it happens.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385116 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The settings saved on the channel for FEATURE()/FEATUREMAP() were only
for that channel.  This patch adds the ability to have these settings
inherited to child channels if you set FEATURE(inherit)=yes.

Closes issue ASTERISK-21306.

Review: https://reviewboard.asterisk.org/r/2415/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385088 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Added a line showing the mapping of "mysql" to res_config_mysql available in add-ons. We used "mysql" as an example driver key in the sample, but didn't show what module it mapped too. Also added a subtitle above the list of keys for driver backends.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@385049 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/2304


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384989 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch sets the protocols container provided by res_http_websocket to NULL
when the module gets unloaded and adds the necessary checks when adding/
removing a websocket protocol. This prevents some FRACKing on an invalid
pointer to the disposed container if a module that uses res_http_websocket is
unloaded after it.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384942 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch does the following:
 * A new Stasis payload has been defined for multi-channel messages. This
   payload can store multiple ast_channel_snapshot objects along with a single
   JSON blob. The payload object itself is opaque; the snapshots are stored
   in a container keyed by roles. APIs have been provided to query for and
   retrieve the snapshots from the payload object.
 * The Dial AMI events have been refactored onto Stasis. This includes dial
   messages in app_dial, as well as the core dialing framework. The AMI events
   have been modified to send out a DialBegin/DialEnd events, as opposed to
   the subevent type that was previously used.
 * Stasis messages, types, and other objects related to channels have been
   placed in their own file, stasis_channels. Unit tests for some of these
   objects/messages have also been written.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384910 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This is the API that binds the Stasis dialplan application to external
Stasis applications. It also adds the beginnings of WebSocket
application support.

This module registers a dialplan function named Stasis, which is used
to put a channel into the named Stasis app. As a channel enters and
leaves the Stasis diaplan application, the Stasis app receives a
'stasis-start' and 'stasis-end' events.

Stasis apps register themselves using the stasis_app_register and
stasis_app_unregister functions. Messages are sent to an application
using stasis_app_send.

Finally, Stasis apps control channels through the use of the
stasis_app_control object, and the family of stasis_app_control_*
functions.

Other changes along for the ride are:
 * An ast_frame_dtor function that's RAII_VAR safe
 * Some common JSON encoders for name/number, timeval, and
   context/extension/priority

Review: https://reviewboard.asterisk.org/r/2361/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384879 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Review: https://reviewboard.asterisk.org/r/2420/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384857 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The initial report was that the "nat" setting in the [general] section was not
having any effect in overriding the default setting.  Upon confirming that this
was happening and looking into what was causing this, it was discovered that
other default settings would not be overriden as well.

This patch works similar to what occurs in build_peer().  We create a temporary
ast_flags structure and using a mask, we override the default settings with
whatever is set in the [general] section.

In the bug report, the reporter who helped to test this patch noted that the
directmedia settings were being overriden properly as well as the nat settings.

This issue is also present in Asterisk 1.8 and a separate patch will be applied
to it.

(issue ASTERISK-21225)
Reported by: Alexandre Vezina
Tested by: Alexandre Vezina, Michael L. Young
Patches:
  asterisk-21225-handle-options-default-prob_v4.diff
						Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2385/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384828 65c4cc65-6c06-0410-ace0-fbb531ad65f3

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384760 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(issue ASTERISK-21151)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384711 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The new inband_on_proceeding option causes Asterisk to assume inband audio
may be present when a PROCEEDING message is received.

Q.931 Section 5.1.2 says the network cannot assume that the CPE side has
attached to the B channel at this time without explicitly sending the
progress indicator ie informing the CPE side to attach to the B channel
for audio.  However, some non-compliant ISDN switches send a PROCEEDING
without the progress indicator ie indicating inband audio is available and
assume that the CPE device has connected the media path for listening to
ringback and other messages.

ASTERISK-17834 which causes this issue was dealing with a non-compliant
network switch.

(closes issue ASTERISK-21151)
Reported by: Gianluca Merlo
Tested by: rmudgett
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Document that you can read/write the 'accountcode' and 'amaflags' on a channel.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384642 65c4cc65-6c06-0410-ace0-fbb531ad65f3

OBJ_PARTIAL_KEY searching a rbtree did not find all possible matches if
the container did not accept duplicates.

Added matching node bias to indicate which matching node is being searched
for: first, last, any.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384616 65c4cc65-6c06-0410-ace0-fbb531ad65f3

func_version.so was being rebuilt every time, because build.h was
changing every build, because of the cleantest dependency that was
added in r384410 to fix parallel make bugs.

Now build.h will only be created if it does not exist, which was the
original behavior of the Makefile.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384518 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Sorry folks. ',' are still greater than '|'.

Thanks for playing along :-)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384514 65c4cc65-6c06-0410-ace0-fbb531ad65f3