We aren't validating that the URI we just parsed is a SIP/SIPS one before
trying to access the user, host, and port members of a possibly uninitialized
structure.

Also update the MessageSend documentation to indicate what 'from' formats are
accepted.

ASTERISK-26484 #close
Reported by: Vinod Dharashive

Change-Id: I476b5cc5f18a7713d0ee945374f2a1c164857d30

POSIX does not require getprotobyname() to be thread safe and some
implementations use static memory which causes issues when multiple
threads are used.

Further, our usage of it today is just to ultimately get IPPROTO_TCP
for calls to setsockopt(). So instead we just use IPPROTO_TCP directly.

Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48

Dereferencing struct ast_autochan.chan without first calling
ast_autochan_channel_lock() is unsafe because the pointer could change at
any time due to a masquerade.  Unfortunately, ast_autochan_channel_lock()
itself uses struct ast_autochan.chan unsafely and can result in a deadlock
if the original channel happens to get destroyed after a masquerade in
addition to the pointer getting changed.

The problem is more likely to happen with v11 and earlier because
masquerades are used to optimize out local channels on those versions.
However, it could still happen on newer versions if the channel is
executing a dialplan application when the channel is transferred or
redirected.  In this situation a masquerade still must be used.

* Added a lock to struct ast_autochan to safely be able to use
ast_autochan.chan while trying to get the channel lock in
ast_autochan_channel_lock().  The locking order is the channel lock then
the autochan lock.  Locking in the other direction requires deadlock
avoidance.

* Fix unsafe ast_autochan.chan usages in app_mixmonitor.c.

* Fix unsafe ast_autochan.chan usages in app_chanspy.c.

* app_chanspy.c: Removed unused autochan parameter from next_channel().

ASTERISK-26867

Change-Id: Id29dd22bc0f369b44e23ca423d2f3657187cc592

Forgetting to indicate an exten is a pattern can cause a crash if the
"pattern" has a character set range.  e.g., "9999[3-5]" The crash is due
to a buffer overwrite because the '-' exten eye-candy wasn't removed as
expected and overran the allocated space.

The buffer overwrite is fixed two ways in this patch.

1) Fix ext_strncpy() to distinguish between pattern and non-pattern
extens.  Now '-' characters are removed when they are eye-candy and not
when they are part of a pattern character set.  Since the function is
private to pbx.c, the return value now returns the number of bytes written
to the destination buffer instead of the strlen() of the final buffer so
the callers that care don't need to add one.

2) Fix callers to ext_strncpy() to supply the correct available buffer
size of the destination buffer.

ASTERISK-26668

Change-Id: I555d97411140e47e0522684062d174fbe32aa84a

* Added additional fields to ast_sdp_options.
* Re-organized ast_sdp.
* Updated field names to correspond to RFC4566 terminology.
* Created allocs/frees for SDP children.
* Created getters/setters for SDP children where appropriate.
* Added ast_sdp_create_from_state.
* Refactored res_sdp_translator_pjmedia for changes.

Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48

This patch demotes the ERROR message that is displayed when a
nonexistent item is removed from the Stasis cache. The genesis of this
demotion is due to chan_sip's realtime peers and their interaction with
Asterisk's core ast_endpoint code, but ostensibly it could happen from
other channel drivers as well.

Since Mark Michelson already did an excellent job of explaining on this
issue, it is quoted here for posterity:

"Internally, when a realtime peer is retrieved, Asterisk creates an
ast_endpoint structure. When that peer is destroyed, the ast_endpoint is
destroyed as well. Part of the destruction of the ast_endpoint involves
clearing the Stasis cache of all information about that endpoint. The
problem here is that the act of creating the ast_endpoint is not enough
to actually put any information in the Stasis cache. Instead, something
has to happen, such as a state change, in order for the Stasis cache to
have any information about that endpoint. When a device registers,
chan_sip creates an ast_endpoint structure, processes the REGISTER, and
then destroys the ast_endpoint. When the ast_endpoint is destroyed,
there is nothing to destroy in the Stasis cache, so an error message is
emitted. When you use rtcachefriends, ast_endpoint structures persist
for the lifetime of the module and so you do not see this error
message."

ASTERISK-25237 #close

Change-Id: I53cebc6b4a897a1ab9564182b75c177780feff70

Change-Id: Icc0dc6e61b2e68d5cdcb74b016b2726a388c7def

This change adds a few things to facilitate stream topology changing:

1. Control frame types have been added for use by the channel driver
to notify the application that the channel wants to change the stream
topology or that a stream topology change has been accepted. They are
also used by the indicate interface to the channel that the application
uses to indicate it wants to do the same.

2. Legacy behavior has been adopted in ast_read() such that if a
channel requests a stream topology change it is denied automatically
and the current stream topology is preserved if the application is
not capable of handling streams.

Tests have also been written which confirm the multistream and
non-multistream behavior.

ASTERISK-26839

Change-Id: Ia68ef22bca8e8457265ca4f0f9de600cbcc10bc9

* say.c Changed 'digits/and' to 'vm-and' for en_GB

ASTERISK-26598 #close

Change-Id: If1b713e5daea6f952b339f139178d292a6c4fcfe

* manager.c:manager_state_cb() Fix potential use of uninitialized hint[]
if a hint does not exist for the requested extension.  Ran into this when
developing a testsuite test.  The AMI event ExtensionStatus came out with
the hint header value containing garbage.  The AMI event PresenceStatus
also had the same issue.

* manager.c:action_extensionstate() no need to completely initialize the
hint[].  Only initialize the first element.

* pbx.c:ast_add_hint() Remove unnecessary assignment.

* chan_sip.c: Eliminate an unneeded hint[] local variable.  We only care
about the return value of ast_get_hint() there.

Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b

* Removed the AST_CHAN_TP_MULTISTREAM tech property.  We now rely
  on read_stream being set to indicate a multi stream channel.
* Added ast_channel_is_multistream convenience function.
* Fixed issue where stream and default_stream weren't being set on
  a frame retrieved from the queue.
* Now testing for NULL being returned from the driver's read or
  read_stream callback.
* Fixed issue where the dropnondefault code was crashing on a
  NULL f.
* Now enforcing that if either read_stream or write_stream are
  set when ast_channel_tech_set is called that BOTH are set.
* Added the unit tests.

ASTERISK-26816

Change-Id: If7792b20d782e71e823dabd3124572cf0a4caab2

This introduces and documents the various states in the state machine.
This also introduces API functions that induce state changes, and places
TODO comments telling what needs to be done in addition to what is
already there. Those TODOs will be replaced with real code in upcoming
changes.

Change-Id: I871c0eb480b4c84d83e91ac5628e7a673e8b89ed

In the event that a cache file is removed out from under us, we should
treat the cache entry as stale and force a refresh.

ASTERISK-26774 #close
Reported by: Igor Gamayunov

Change-Id: I3b1bd0c999d59d18664ef73a29823bc5b431dc52

... and clean them both up on uninstall.

We've fixed the issue where 'make install' was installing to
/usr/lib on 64-bit systems that use /usr/lib64.  Now we need
to clean up the remnants in /usr/lib.

* 'make install' now prints a warning if DESTDIR/ASTLIBDIR
  contains 'lib64' and libasterisk* shared libraries or modules
  are also found in DESTDIR/ASTLIBDIR with 'lib64' transformed
  to 'lib'.

* 'make uninstall' ALWAYS cleans up both DESTDIR/ASTLIBDIR and
  DESTDIR/ASTLIBDIR with 'lib64' transformed to 'lib'.

ASTERISK-26705

Change-Id: I6edddeb3c07a51e7c7ba7cac3c05e4bf3ec3f01f

This change introduces an ast_read_stream function and callback in
the channel technology which allows reading frames from all streams
and not just the default streams.

The stream number has also been added to frames. This is to allow the
case where frames are queued onto the channel instead of being read
directly from the driver.

This change does impose a restriction on reading though: a chain of
frames can only contain frames from the same stream.

ASTERISK-26816

Change-Id: I5d7dc35e86694df91fd025126f6cfe0453aa38ce

On some platforms a multiarch approach is used for libraries.
The build system does not take this into account and still
places libraries into the lib directory if no --libdir is
specified to configure. On initial startup this results in
libasteriskssl.so not being found, as it is not in the multiarch
lib directory.  To make matters worse, options were being passed
to ldconfig on both Linux and FreeBSD that actually prevented
the rebuild of the cache.

 * Fedora has a /usr/share/config.site that automatically tells
   autoconf to use /usr/lib64 but CentOS does not. This logic was
   copied to configure.ac and modified so systems like Ubuntu,
   which still use /usr/lib for 64-bit systems, aren't affected.

Now that we have them in the correct directory...

In order for the system loader to find libasteriskssl and
libasteriskpj, one of 3 things has to happen...

  - The linker cache must be rebuilt including the directory
    where the libasterisk* libraries were installed.  Only root
    can rebuild the cache.  This was busted.
  - We have to link the asterisk binary with an rpath pointing
    to the directrory where the libasterisk* libraries were
    installed.  This makes things very complicated and will happen
    over the collective dead bodies of everyone who's had to
    package a distribution with an rpath.
  - Finally, you can start asterisk with LD_LIBRARY_PATH set to the
    directrory where the libasterisk* libraries were installed.

There are no other options. So...

 * The invokation of ldconfig has been moved from main/Makefile
   to ASTTOPDIR/Makefile, the options have been removed, and
   DESTDIR/ASTLIBDIR appended.  If you aren't root, you will be
   warned after the "Asterisk Installation Compete" banner that
   you must re-run 'make install' as root, manually run
   'ldconfig DESTDIR/ASTLIBDIR' as root, or run asterisk with
   LD_LIBRARY_PATH.

ASTERISK-26705

Change-Id: I2a64b7c33a7d3e9bde20f47e3d3ab771977af982

This change adds an ast_write_stream function which allows
writing a frame to a specific media stream. It also moves
ast_write() to using this underneath by writing media
frames provided to it to the default streams of the channel.
Existing functionality (such as audiohooks, framehooks, etc)
are limited to being applied to the default stream only.

Unit tests have also been added which test the behavior of
both non-multistream and multistream channels to confirm that
the write() and write_stream() callbacks are invoked
appropriately.

ASTERISK-26793

Change-Id: I4df20d1b65bd4d787fce0b4b478e19d2dfea245c

Adds binaural synthesis to bridge_softmix (via convolution using libfftw3).
Binaural synthesis is conducted at 48kHz.
For a conference, only one spatial representation is rendered.
The default rendering is applied for mono-capable channels.

ASTERISK-26292

Change-Id: Iecdb381b6adc17c961049658678f6219adae1ddf

This reverts commit 28c8e4f5.

Change-Id: Ie2e1aaf61fd49045994974a4581545ac8348fe4c

The "core show channel" CLI command will now output the streams
present on the channel with their details.

ASTERISK-26811

Change-Id: I9c95b57aa09415005f0677a1949a0feb07e4987a

This establishes the basic allocation/destruction of an SDP state
object, plus some of the simpler getter methods involved. Subsequent
tasks will deal with adding a state machine, creating SDPs from
capabilities and options, and merging SDPs into a joint SDP.

Change-Id: Ie3757ce186f04b65e9d1883f5aace53f24e53709

On some platforms a multiarch approach is used for libraries.
The build system does not take this into account and still
places libraries into the lib directory if no --libdir is
specified to configure. On initial startup this results in
libasteriskssl.so not being found, as it is not in the multiarch
lib directory.

This change does the minimally invasive thing and executes
ldconfig so that the libraries in the lib directory are found
and their location cached. By doing so Asterisk starts up fine.

If DESTDIR is specified, however, the old logic is executed as
the install process may not have permission to alter the ldconfig
cache.

ASTERISK-26705

Change-Id: If4eca46ac510c6fea5568256280ffdb3888d7bb4

This reverts commit 8851c3e0.

Change-Id: I124380be5e3bd57da978428a2a93604336ccd0db

* Fix tcptls_session ref and fd leak in ast_tcptls_server_root().

Change-Id: I0ddf01cd3c10d3b6666d7bf68d4e206a37f4fbdb

When AMI encounters an error at the beginning of a session, it would
explicitly call ast_iostream_close() on its tcptls session's iostream.
It then would jump to a label where it would shut down the tcptls
session instance. The tcptls session instance would again attempt to
close the iostream.

Under normal circumstances, this might go by unnoticed. However, when
MALLOC_DEBUG is enabled, all fields on the iostream get set to
0xdeaddead when the iostream is freed. Thus a second call to
ast_iostream_close() after the iostream has been freed would reslt in an
attempt to call SSL_shutdown on 0xdeaddead, which would crash and burn
horribly.

The fix here is to not directly close the iostream from the dangerous
scenarios. The specific scenarios are:
* Exceeding the configured authlimit
* Failing to build a mansession on a new connection

Change-Id: I908f98d516afd5a263bd36b072221008a4731acd

This creates the following:
* Asterisk's internal representation of an SDP
* An API for translating SDPs from one format to another
* An implementation of a translator for PJMEDIA

Change-Id: Ie2ecd3cbebe76756577be9b133e84d2ee356d46b

This is step one of adding an SDP API: defining some
configurable settings for SDPs. This is based on options
that are currently supported in Asterisk.

Change-Id: I1ede91aafed403b12a9ccdfb91a88389baa7e5d7

On some platforms a multiarch approach is used for libraries.
The build system does not take this into account and still
places libraries into the lib directory if no --libdir is
specified to configure. On initial startup this results in
libasteriskssl.so not being found, as it is not in the multiarch
lib directory.

This change does the minimally invasive thing and executes
ldconfig so that the libraries in the lib directory are found
and their location cached. By doing so Asterisk starts up fine.

ASTERISK-26705

Change-Id: I6d30b6427e9d5e69470e11327c7ff203fa7da519

To be consistent with sdp implementation.

Change-Id: I714e300939b4188f58ca66ce9d1e84b287009500

ASTERISK-26794 #close

Change-Id: I9cbc3b6b6a8aab590f5ccde9c262a98e4d5253a1

Adds topology set and get to channel.

ASTERISK-26790

Change-Id: Ic379ea82a9486fc79dbd8c4d95c29fa3b46424f4

This change adds unit tests for the various API calls relating
to stream topologies. This includes creation, destruction,
inspection, and manipulation.

Through this a few bugs were uncovered in the implementation:

1. Creating a topology using a format capabilities would fail as
the code considered a return value of 0 from the append stream
function to indicate an error which is incorrect.

2. Not all functions which placed a stream into a topology
set the position on the stream itself.

3. Appending a stream would cause a frack if the position
provided was the last one. This occurred because the existing
stream was queried but the index was outside of what the
vector was currently at for size.

ASTERISK-26786

Change-Id: Id5590e87c8a605deea1a89e53169a9c011d66fa0

* app_minivm: Use built-in completion facilities to complete optional
arguments.

* app_voicemail: Use built-in completion facilities to complete
optional arguments.

* app_confbridge: Add missing colons after 'Usage' text.

* chan_alsa: Use built-in completion facilities to complete optional
arguments.

* chan_sip: Use built-in completion facilities to complete optional
arguments. Add completions for 'load' for 'sip show user', 'sip show
peer', and 'sip qualify peer.'

* chan_skinny: Correct and extend completions for 'skinny reset' and
'skinny show line.'

* func_odbc: Correct completions for 'odbc read' and 'odbc write'

* main/astmm: Use built-in completion facilities to complete arguments
for 'memory' commands.

* main/bridge: Correct completions for 'bridge kick.'

* main/ccss: Use built-in completion facilities to complete arguments
for 'cc cancel' command.

* main/cli: Add 'all' completion for 'channel request hangup.' Correct
completions for 'core set debug channel.' Correct completions for 'core
show calls.'

* main/pbx_app: Remove redundant completions for 'core show
applications.'

* main/pbx_hangup_handler: Remove unused completions for 'core show
hanguphandlers all.'

* res_sorcery_memory_cache: Add completion for 'reload' argument of
'sorcery memory cache stale' and properly implement.

Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca

This change adds the media stream topology definition and API for
accessing and using it.

Some refactoring of the stream was also done.

ASTERISK-26786

Change-Id: Ic930232d24d5ad66dcabc14e9b359e0ff8e7f568

The ast_waitfor_nandfds operation will manipulate the flags
of channels passed in. This was previously done without
the channel lock being held. This could result in incorrect
values existing for the flags if another thread manipulated
the flags at the same time.

This change locks the channel during flag manipulation.

ASTERISK-26788

Change-Id: I2c5c8edec17c9bdad4a93291576838cb552ca5ed

In Asterisk 11, if the 'Originate' AMI command failed to connect the provided
Channel while in extension mode, a 'failed' extension would be looked up and
run. This was, I believe, unintentionally removed in 51b6c496. This patch
restores that behavior.

This also adds an enum for the various 'synchronous' modes in an attempt to
make them meaningful.

ASTERISK-26115 #close
Reported by: Nasir Iqbal

Change-Id: I8afbd06725e99610e02adb529137d4800c05345d

We shouldn't unlock the channel after starting a snapshot staging because
another thread may interfere and do its own snapshot staging.

* app_dial.c:dial_exec_full() made hold the channel lock while setting up
the outgoing channel staging.  Made hold the channel lock after the called
party answers while updating the caller channel staging.

* chan_sip.c:sip_new() completed the channel staging on off-nominal exit.
Also we need to use ast_hangup() instead of ast_channel_unref() at that
location.

* channel.c:__ast_channel_alloc_ap() added a comment about not needing to
complete the channel snapshot staging on off-nominal exit paths.

* rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel
locks while staging the channels for the stats channel variables.

Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a

This change adds the media stream definition and API for
accessing and using it. Unit tests have also been written
which exercise aspects of the API.

ASTERISK-26773

Change-Id: I3dbe54065b55aaa51f467e1a3bafd67fb48cac87

When performing an SRV lookup using the ast_srv_lookup function it
did not properly handle the situation where 0 records are returned.
If this happened it would wrongly assume that at least one record
was present.

This change fixes the code so it will exit early if an error occurs
or if 0 records are returned.

ASTERISK-26772
patches:
  srv_lookup.patch submitted by nappsoft (license 6822)

Change-Id: I09b19081c74e0ad11c12bf54a257243b1bcb2351

In ari.conf, when setting the option channelvars, every Stasis channel
snapshot would create a list of variable/value that would not be freed
when the snapshot is freed, resulting in a often-recurring memory
leak.

ASTERISK-26767 #close

Change-Id: Ia37dd9d68063d7f879193df02ede293e5ded716d