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

This patch adds a new features to the endpoint identifier module,
'match_header'. When set, inbound requests are matched by a provided SIP
header: value pair. This option works in conjunction with the existing
'match' configuration option, such that if any 'match*' attribute
matches an inbound request, the request is associated with the specified
endpoint.

Since this module now identifies by more than just IP address,
appropriate renaming of the module and/or variables can be done in a
non-release branch.

ASTERISK-26863 #close

Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453
(cherry picked from commit 30f52d79)

In some scenarios, such as when there may not be a terminal (such as
inside a Docker container), curl will apparently direct the progress bar
to stdout. This can cause extra data to be appended to a file curl'd
down to stdout, resulting in md5 verification failures.

This patch removes the progress bar, and tells curl to download the file
silently.

ASTERISK-26872 #close

Change-Id: Ie860b020f627d4372b3e7ce9453de5faafeebe6c

This patch updates the documenation in hep.conf.sample to better specify
how the various HEP modules interact.

ASTERISK-26717 #close

Change-Id: I337fb742a89e3ec5edc7fc7a7a0295218d841124

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

Tabs > spaces. Always.

Change-Id: I899ff662361c7ab0327173bd7851a67b53dd65f1

When querying for PJSIP specific information using the dialplan
function CHANNEL() it is possible that the underlying session
will no longer have a channel associated with it. This is
most likely to occur when the RTCP HEP module attempts to get
the channel name. If this happens then a crash will occur.

This change just adds a check that the channel exists on the
session before querying it.

ASTERISK-26857

Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01

Bundled pjproject should now only rebuild if one of the menuselect
"Compiler Flags" options changes.

Change-Id: If114a2e16b9e77af371a600d6a5e197bbf28fe43

* cli_commands.c Fixed CLI output

ASTERISK-26822 #close

Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01

* res_musiconhold.c: Ensure the general section is not treated as
a moh class.

ASTERISK-26353 #close

Change-Id: Ia3dbd11ea2b43ab3e6c820a9827811dd24bea82d

Change-Id: Icc0dc6e61b2e68d5cdcb74b016b2726a388c7def

Set a variable on the channel that indicates which attempt number we
are currently performing to allow for attempt-specific behavior.

ASTERISK-26568 #close
Reported by: Roman Shubovich

Change-Id: Iacd7e8d43b0ed5b6cb021c62f41f1a1f5733dd89

This change adds a PJSIP patch (which has been contributed upstream)
to allow the registration of IPv6 transport types.

Using this the res_pjsip_transport_websocket module now registers
an IPv6 Websocket transport and uses it for the corresponding
traffic.

ASTERISK-26685

Change-Id: Id1f9126f995b31dc38db8fdb58afd289b4ad1647

* apps/app_voicemail.c fromstring field added to mailbox which will
override the global fromstring if set.

ASTERISK-24562 #close

Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe

When doing some WebRTC testing, I found that the websocket would
disconnect whenever I attempted to place a call into Asterisk. After
looking into it, I pinpointed the problem to be due to the iostreams
change being merged in.

Under certain circumstances, a call to ast_iostream_read() can return a
negative value. However, in this circumstance, the websocket code was
treating this negative return as if it were a partial read from the
websocket. The expected length would get adjusted by this negative
value, resulting in the expected length being too large.

This patch simply adds an if check to be sure that we are only updating
the expected length of a read when the return from a read is positive.

ASTERISK-26842 #close
Reported by Mark Michelson

Change-Id: Ib4423239828a013d27d7bc477d317d2f02db61ab

When receiving a 422 response, the invitestate variable must be reset to
INV_CALLING.

ASTERISK-26841

Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099

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

Per the linked issue, we aren't checking the buffer filled by fgets()
to determine if it contains a newline, so we will fail to correctly
parse the trailing portion of a long line.

This patch increases the buffer size from 256 to 1024, and skips any
line that exceeds that length, logging a warning in the process.

ASTERISK-17067 #close
Reported by: Dave Olszewski

Change-Id: I51bcf270c1b4347ba05b43f18dc2094c76f5d7b0

* 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

According to the RFC[1] WSS should only be used in the Via header
for secure Websockets.

* Use WSS in Via for secure transport.

* Only register one transport with the WS name because it would be
ambiguous.  Outgoing requests may try to find the transport by name and
pjproject only finds the first one registered.  This may mess up unsecure
websockets but the impact should be minimal.  Firefox and Chrome do not
support anything other than secure websockets anymore.

* Added and updated some debug messages concerning websockets.

* security_events.c: Relax case restriction when determining security
transport type.

* The res_pjsip_nat module has been updated to not touch the transport
on Websocket originating messages.

[1] https://tools.ietf.org/html/rfc7118

ASTERISK-26796 #close

Change-Id: Ie3a0fb1a41101a4c1e49d875a8aa87b189e7ab12

* 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