This change allows chan_pjsip to be given an AST_FRAME_RTCP
containing REMB feedback and pass it to res_rtp_asterisk.
Once res_rtp_asterisk receives the frame a REMB RTCP feedback
packet is constructed with the appropriate contents and sent
to the remote endpoint.

ASTERISK-27776

Change-Id: Ic53f821c1560d8924907ad82c4d9c0bc322b38cd

This change extends the existing AST_FRAME_RTCP frame type to be
able to contain additional RTCP message types, such as feedback
messages. The payload type is contained in the subclass which allows
knowing what is in the frame itself.

The RTCP feedback message type is now handled and REMB[1] messages
are raised with their containing information.

This also fixes a bug where all feedback messages were triggering
video updates instead of just FIR and FUR.

Finally RTCP frames are now passed up through the Asterisk core to
what is handling the channel, mapped appropriately in the case of
bridging, and written to an outgoing stream. Since RTCP frames are
on a per-stream basis this is only done on multistream capable
channels.

[1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03

ASTERISK-27758
ASTERISK-26366

Change-Id: I680da0ad8d5059d5e9655d896fb9d92e9da8491e

In the script ./configure, AST_EXT_LIB_CHECK checks for external libraries. Some
libraries do not specify all their dependencies and require additional shared
libraries. In AST_EXT_LIB_CHECK, this is the fifth parameter. However, if a
library is specified there, it must exist on the platform, because ./configure
tries to compile/link/execute a small app using those statements. For example,
the library libdl.so is Linux specific and does not exist on BSD-like platforms.

Furthermore, no supported platform/version was found, which still (ever?)
requires those additional libraries. Therefore, they were simply removed.

Finally, this change adds the error code ESTRPIPE to the channel driver
chan_alsa for those platforms which lack it, again for example NetBSD.

ASTERISK-27720

Change-Id: I3b21f2135f6cbfac7590ccdc2df753257f426e0b

* acl (named_acl.c)
* cdr
* cel
* ccss
* dnsmgr
* dsp
* enum
* extconfig (config.c)
* features
* http
* indications
* logger
* manager
* plc
* sounds
* udptl

These modules are now loaded at appropriate time by the module loader.
Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so
the module loader will abort startup on failure of these modules.

Some of these modules are still initialized or shutdown from outside the
module loader.  logger.c is initialized very early and shutdown very
late, manager.c is initialized by the module loader but is shutdown by
the Asterisk core (too much uses it without holding references).

Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f

Checking option_debug directly is incorrect as it ignores file/module
specific debug settings.  This system-wide change replaces nearly all
direct checks for option_debug with the DEBUG_ATLEAST macro.

Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0

The "ptime" SDP parameter received in a SIP response was not honoured.
Moreover, in the abscence of this "ptime" parameter, locally configured
framing was lost during response processing.

This patch systematically stores the framing information in the
ast_rtp_codecs structure, taking it from the response or from the
configuration as appropriate.

ASTERISK-27674

Change-Id: I828a6a98d27a45a8afd07236a2bd0aa3cbd3fb2c

ASTERISK-27714
Reported by: John Nemeth

Change-Id: I1b84a89315a5f61222123d21bf35c59224da8990

When constructing a dialog-info+xml NOTIFY message a ringing channel
is found if the state is ringing and further information is placed into
the message. Due to the migration to the Stasis message bus this did
not always work as expected.

This change raises a second ringing event in such a way to guarantee
that the event is received by chan_sip and another lookup is done to
find the ringing channel.

ASTERISK-24488

Change-Id: I547a458fc59721c918cb48be060cbfc3c88bcf9c

Check if initreq data string exists before using it when processing a
CANCEL request.

ASTERISK-27666

Change-Id: Id1d0f0fa4ec94e81b332b2973d93e5a14bb4cc97

It seems that the ALSA backend of PortAudio doesn't know how to both
read and write at the same time by adding a per-device mutex.

FIXME: currently only a draft version. Need to either auto-detect
we work with the ALSA backend or add an extra configuration option
to use this mutex.

ASTERISK-27426 #close

Change-Id: I635eacee45f5413faa18f5a3b606af03b926dacb

This removes references that are no longer needed due to automatic
references created by module dependencies.

In addition this removes most calls to ast_module_check as they were
checking modules which are listed as dependencies.

Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e

This patch fix chan_unistim hold functions to correctly support
hold function in different states possible in case of multiple lines
established on the phone

ASTERISK-26596 #close

Change-Id: Ib1e04e482e7c8939607a42d7fddacc07e26e14d4

* Declare 'requires' and 'enhances' text fields on module info structure.
* Rename 'nonoptreq' to 'optional_modules'.
* Update doxygen comments.

Still need to investigate dependencies among modules I cannot compile.

Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf

Before getting the file descriptor for an iostream check
that it is present.

ASTERISK-27534

Change-Id: Ie0aa1394007a37c30e337ea1176a6fb3a63bc99c

Per RFC 5245, the foundation specified with an ICE candidate can be up
to 32 characters but we are only allowing for 31.

ASTERISK-27498 #close
Reported by: Michele Prà

Change-Id: I05ce7a5952721a76a2b4c90366168022558dc7cf

This is the old ASTOBJ macro's which are no longer used except by the
deprecated netsock.c.  Move it to the chan_iax2 include folder so it
does not get used elsewhere.

Change-Id: I7e4ae96678b36b9f41d3cae14b167f110eb5d349

Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0

Fix instances of:
* Retreive
* Recieve
* other then
* different then
* Repeated words ("the the", "an an", "and and", etc).
* othterwise, teh

ASTERISK-24198 #close

Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31

In change_redirecting_information variables we use ast_strlen_zero to
see if a value should be saved.  In the case where the value is not NULL
but is a zero length string we leaked.

handle_response_subscribe leaked a reference to the ccss monitor
instance.

Change-Id: Ib11444de69c3d5b2360a88ba2feb54d2c2e9f05f

Some variables are set and never changed, making them constant.  This
means that code in the 'false' block of the conditional is unreachable.

In chan_skinny and res_config_ldap I used preprocessor directive `#if 0`
as I'm unsure if the unreachable code could be enabled in the future.

Change-Id: I62e2aac353d739fb3c983cf768933120f5fba059

chan_console supports multiple devices but the CLI only works on a
single device. 'console set active' selects this device.

Sadly that CLI picks the wrong command-line parameter and will only
work for a device called 'active'.

ASTERISK-27490 #close

Change-Id: I2f0e5fe63db19845bee862575b739360797dc73d

This moves netsock.c / netsock.h to the chan_iax2 module.  netsock.h has
been marked deprecated since 13.0.0, chan_iax2 is the only remaining
user.

Change-Id: I28c6578043bac18de5ea608e136acec4f83d5dd3

Attempting to dial PJSIP/endpoint when the endpoint doesn't exist and
disable_multi_domain=no results in a misleading empty endpoint name
message.  The message should say the endpoint was not found.

* Added missing endpoint not found message.

* Added more information to the empty endpoint name msgs if available.

* Eliminated RAII_VAR in request().

Change-Id: I21da85ebd62dcc32115b2ffcb5157416ebae51e4

Log a message to security events when an INVITE is received to an
invalid extension.

ASTERISK-25869 #close

Change-Id: I0da40cd7c2206c825c2f0d4e172275df331fcc8f

Remove nearly all use of regex from ACO users.  Still remaining:
* app_confbridge has a legitamate use of option name regex.
* ast_sorcery_object_fields_register is implemented with regex, all
  callers use simple prefix based regex.  I haven't decided the best
  way to fix this in both 13/15 and master.

Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b

A patch for sending in-dialog SIP NOTIFY message
with "SIPnotify" AMI action.

ASTERISK-27461

Change-Id: I5797ded4752acd966db6b13971284db684cc5ab4

This is a partial fix for ASTERISK~25817 but does not address the
comments regarding RFC 5626.

Change-Id: I227e2d10c0035bbfa1c6e46ae2318fd1122d8420

Stripping the DNID in a SIP dial string can result in attempting to call
the argument parsing macros on an empty string, causing a crash.

ASTERISK-26131 #close
Reported by: Dwayne Hubbard
Patches:
	dw-asterisk-master-dnid-crash.patch (license #6257) patch
	uploaded by Dwayne Hubbard

Change-Id: Ib84c1f740a9ec0539d582b09d847fc85ddca1c5e

This patch does three things associated with the initial incoming INVITE
request URI.

1) Add access to the full initial incoming INVITE request URI.

2) We were not setting DNID on incoming PJSIP channels.  The DNID is the
user portion of the initial incoming INVITE Request-URI.  The value is
accessed by reading CALLERID(dnid).

3) Fix CHANNEL(pjsip,target_uri) documentation.

* The initial incoming INVITE request URI is now available using
CHANNEL(pjsip,request_uri).

* Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the
initial incoming INVITE request URI user portion.

* CHANNEL(pjsip,target_uri) now correctly documents that the target URI is
the contact URI.

* Refactored print_escaped_uri() out of channel_read_pjsip() to handle
pjsip_uri_print() error condition when the buffer is too small.

ASTERISK-27478

Change-Id: I512e60d1f162395c946451becb37af3333337b33

There are many places in the code base where we ignore the return value
of fcntl() when getting/setting file descriptior flags. This patch
introduces a convenience function that allows setting or clearing file
descriptor flags and will also log an error on failure for later
analysis.

Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7

The SuccessfulAuth using_password field was declared as a pointer to a
uint32_t when the field was later read as a uint32_t value.  This resulted
in unnecessary casts and a non-portable field value reinterpret in
main/security_events.c:add_json_object().  i.e., It would work on a 32 bit
architecture but not on a 64 bit big endian architecture.

Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935

Previously, peers connected via TCP (or TLS) were matched by ignoring their
source port. One cannot say anything when protocol:IP:port match, yes (see
<http://stackoverflow.com/q/3329641>). However, when the ports do not match, the
peers do not match as well.

This change allows two peers connected to an Asterisk server via TCP (or TLS)
behind a NAT (= same source IP address) to be differentiated via their port as
well.

ASTERISK-27457
Reported by: Stephane Chazelas

Change-Id: Id190428bf1d931f2dbfd4b293f53ff8f20d98efa

chan_skinny creates a new thread for each new session.  In trying
to be a good cleanup citizen, the threads are joinable and the
unload_module function does a pthread_cancel() and a pthread_join()
on any sessions that are active at that time.  This has an
unintended side effect though. Since you can call pthread_join on a
thread that's already terminated, pthreads keeps the thread's
storage around until you explicitly call pthread_join (or
pthread_detach()).   Since only the module_unload function was
calling pthread_join, and even then only on the ones active at the
tme, the storage for every thread/session ever created sticks
around until asterisk exits.

* A thread can detach itself so the session_destroy() function
  now calls pthread_detach() just before it frees the session
  memory allocation.  The module_unload function still takes care
  of the ones that are still active should the module be unloaded.

ASTERISK-27452
Reported by: Juan Sacco

Change-Id: I9af7268eba14bf76960566f891320f97b974e6dd
(cherry picked from commit 8f5dff54)

ASTERISK-27434

Change-Id: Iaeed89b4fa05d94c5f0ec2d3b7cd6e93d2d5a8f7

* Balanced the session->inv_session refs on answer failure.

Change-Id: I33542d639d37e692cb46550b972a5fcfc3b804b8

The media frame cache gets in the way of finding use after free errors of
media frames.  Tools like valgrind and MALLOC_DEBUG don't know when a
frame is released because it gets put into the cache instead of being
freed.

* Added the "cache_media_frames" option to asterisk.conf.  Disabling the
option helps track down media frame mismanagement when using valgrind or
MALLOC_DEBUG.  The cache gets in the way of determining if the frame is
used after free and who freed it.  NOTE: This option has no effect when
Asterisk is compiled with the LOW_MEMORY compile time option enabled
because the cache code does not exist.

To disable the media frame cache simply disable the cache_media_frames
option in asterisk.conf and restart Asterisk.

Sample asterisk.conf setting:
[options]
cache_media_frames=no

ASTERISK-27413

Change-Id: I0ab2ce0f4547cccf2eb214901835c2d951b78c00

Change-Id: I3f9dd3c31bd582e54a30381500077de2319d8cc3

This mimics the behavior of Chrome and Firefox and creates an ephemeral
X.509 certificate for each DTLS session.

Currently, the only supported key type is ECDSA because of its faster
generation time, but other key types can be added in the future as
necessary.

ASTERISK-27395

Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4

This adds menuselect dependencies for modules that use symbols of other
modules.

ASTERISK-27390

Change-Id: Ia2d2849f5b87a72af7324a82edc3f283eafb5385

When chan_sip receives a SUBSCRIBE request with no "Expires" header it
processes the request as an unsubscribe.  This is incorrect, per RFC3264
when the "Expires" header is missing a default expiry should be used.

ASTERISK-18140

Change-Id: Ibf6dcd4fdd07a32c2bc38be1dd557981f08188b5