gethostbyname() and gethostbyname_r() are deprecated in favor of
getaddrinfo() which we use in the ast_sockaddr family of functions.

ASTERISK-29819 #close

Change-Id: Ie277c0ef768d753b169c121ef570a71665692ab7

Attempting to access ${CHANNEL(ruri)} in a pre-dial handler before
initiating an outgoing call will cause Asterisk to crash. This is
because a null field is accessed, resulting in an offset from null and
subsequent memory access violation.

Since RURI is not guaranteed to exist, we now check if the base
pointer is non-null before calculating an offset.

ASTERISK-29772

Change-Id: Icd3b02f07256bbe6615854af5717074087b95a83

ASTERISK-29779

Change-Id: If338163488498f65fa7248b60e80299c0a928e4b

Since Doxygen 1.8.16, a special comment block is required. Otherwise
(pure C comment), the group command is ignored. Additionally, several
unbalanced group commands were fixed.

ASTERISK-29732

Change-Id: I4687857b9d56e6f44fd440b73af156691660202e

Most examples in the XML documentation use the
example tag to demonstrate examples, which gets
parsed specially in the Wiki to make it easier
to follow for users.

This fixes a few modules to use the example
tag instead of vanilla para tags to bring them
in line with the standard syntax.

ASTERISK-29777 #close

Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b

ASTERISK-29764

Change-Id: I6e5466cce03e25695c5c7d8b68c305184dcf5375

ASTERISK-29762

Change-Id: Ia8811ac12b93ff8c18164699c6fbc604cb0a23f7

ASTERISK-29737

Change-Id: I282003cc553989fd5c19ceeac9e478fa4ee06cec

Correct typos of the following word families:

appease
permanently
overriding
residue
silliness
extension
channels
globally
reference
japanese
group
coordinate
registry
information
inconvenience
attempts
cadence
payloads
presence
provisioning
mimics
behavior
width
natively
syslabel
not owning
unquelch
mostly
constants
interesting
active
unequipped
brodmann
commanding
backlogged
without
bitstream
firmware
maintain
exclusive
practically
structs
appearance
range
retransmission
indication
provisional
associating
always
whether
cyrillic
distinctive
components
reinitialized
initialized
capability
switches
occurring
happened
outbound

ASTERISK-29714

Change-Id: Ife52ee89cd2170b684fa651ca72b1cb911a57339

Fixes compiler warning caused by a truncated copy of the ANI2 into a
buffer of size 10. This could prevent the null terminator from being
copied if the copy value exceeds the size of the buffer. This increases
the buffer size to 101 to ensure there is no way for truncation to occur.

ASTERISK-29702 #close

Change-Id: Ief9052212952840fa44de6463b8699fdb3e163d0

Historically, the dial syntax for IAX2 has held that
an outkey (used only for RSA authenticated calls)
and a secret (used only for plain text and MD5 authenticated
calls, historically) were mutually exclusive, and thus
the same position in the dial string was used for both
values.

Now that encryption is possible with RSA authentication,
this poses a limitation, since encryption requires a
secret and RSA authentication requires an outkey. Thus,
the dial syntax is extended so that both a secret and
an outkey can be specified.

The new extended syntax is backwards compatible with the
old syntax. However, a secret can now be specified after
the outkey, or the outkey can be specified after the secret.
This makes it possible to spawn an encrypted RSA authenticated
call without a corresponding peer being predefined in iax.conf.

ASTERISK-29707 #close

Change-Id: I1f8149313ed760169d604afbb07720a8b07dd00e

test_voicemail_api: Use empty char* for empty_msg_ids.
chan_skinny: Fix size of calledParty to be maximum extension.
menuselect: Change Makefile to stop deprecated warnings. Added comments
test_linkedlist: 'bogus' variable was manually allocated from a macro
and the test fails if this happens but the compiler couldn't 'see' this
and returns a warning. memset to all 0's after allocation.
chan_ooh323: Fixed various indentation issues that triggered misleading
 indentation warnings.

ASTERISK-29682
Reported by: George Joseph

Change-Id: If4fe42222c8444dc16828a42731ee53b4ce5cbbe

Adds support for encryption to RSA-authenticated
calls. Also prevents crashes if an RSA IAX2 call
is initiated to a switch requiring encryption
but no secret is provided.

ASTERISK-20219

Change-Id: I18f1f9d7c59b4f9cffa00f3b94a4c875846efd40

Adds parsing of ANI II digits (Originating
Line Information) to PJSIP, on par with
what currently exists in chan_sip.

ASTERISK-29472

Change-Id: Ifc938a7a7d45ce33999ebf3656a542226f6d3847

Adds an information element for ANI2 so that
Originating Line Information can be transmitted
over IAX2 channels.

ASTERISK-29605 #close

Change-Id: Iaeacdf6ccde18eaff7f776a0f49fee87dcb549d2

This changeset is intended to address compatibility issues encountered
when interfacing Asterisk to electromechanical telephone switches that
implement ANI-B, ANI-C, or ANI-D.

In particular the behaviours that this impacts include:

 - FGC-CAMA did not work at all when using MF signaling. Modified the
   switch case block to send calls to the correct part of the
   signaling-handling state machine.

 - For FGC-CAMA operation, the delay between called number ST and
   second wink for ANI spill has been made configurable; previously
   all calls were made to wait for one full second.

 - After the ANI spill, previous behavior was to require a 'ST' tone
   to advance the call.  This has been changed to allow 'STP' 'ST2P'
   or 'ST3P' as well, for compatibility with ANI-D.

 - Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable.

 - For calls with an ANI failure, No. 1 Crossbar switches will send
   forward a single-digit failure code, with no calling number digits
   and no ST pulse to terminate the spill.  I've made the ANI timeout
   configurable so to reduce dead air time on calls with ANI fail.

 - ANI info digits configurable.  Modern digital switches will send 2
   digits, but ANI-B sends only a single info digit.  This caused the
   ANI reported by Asterisk to be misaligned.

 - Changed a confusing log message to be more informative.

ASTERISK-29518

Change-Id: Ib7e27d987aee4ed9bc3663c57ef413e21b404256

ASTERISK-20339 #close

Change-Id: I36f364aaa1971241d8f3ea1a5909b463d185a2d5

app_meetme is deprecated in 19, to be removed in 21.
app_osplookup is deprecated in 19, to be removed in 21.
chan_alsa is deprecated in 19, to be removed in 21.
chan_mgcp is deprecated in 19, to be removed in 21.
chan_skinny is deprecated in 19, to be removed in 21.
res_pktccops is deprecated in 19, to be removed in 21.
cdr_mysql was deprecated in 1.8, to be removed in 19.
app_mysql was deprecated in 1.8, to be removed in 19.
app_ices was deprecated in 16, to be removed in 19.
app_macro was deprecated in 16, to be removed in 21.
app_fax was deprecated in 16, to be removed in 19.
app_url was deprecated in 16, to be removed in 19.
app_image was deprecated in 16, to be removed in 19.
app_nbscat was deprecated in 16, to be removed in 19.
app_dahdiras was deprecated in 16, to be removed in 19.
cdr_syslog was deprecated in 16, to be removed in 19.
chan_oss was deprecated in 16, to be removed in 19.
chan_phone was deprecated in 16, to be removed in 19.
chan_sip was deprecated in 17, to be removed in 21.
chan_nbs was deprecated in 16, to be removed in 19.
chan_misdn was deprecated in 16, to be removed in 19.
chan_vpb was deprecated in 16, to be removed in 19.
res_config_sqlite was deprecated in 16, to be removed in 19.
res_monitor was deprecated in 16, to be removed in 21.
conf2ael was deprecated in 16, to be removed in 19.
muted was deprecated in 16, to be removed in 19.

ASTERISK-29548
ASTERISK-29549
ASTERISK-29550
ASTERISK-29551
ASTERISK-29552
ASTERISK-29553
ASTERISK-29554
ASTERISK-29555
ASTERISK-29557
ASTERISK-29558
ASTERISK-29559
ASTERISK-29560
ASTERISK-29561
ASTERISK-29562
ASTERISK-29563
ASTERISK-29564
ASTERISK-29565
ASTERISK-29566
ASTERISK-29567
ASTERISK-29568
ASTERISK-29569
ASTERISK-29570
ASTERISK-29571
ASTERISK-29572
ASTERISK-29573
ASTERISK-29574

Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131

If chan_iax2 received a packet with an unsupported media format, for
example vp9, then it would set the frame's format to NULL. This could
then result in a crash later when an attempt was made to access the
format.

This patch makes it so chan_iax2 now ignores/drops frames received
with unsupported media format types.

ASTERISK-29392 #close

Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1

Some ATAs send hook flash events as application/hook-flash, rather than a DTMF
event. Now, we also recognize hook-flash as a flash event.

ASTERISK-29370

Change-Id: I1c3b82a040dff3affcd94bad8ce33edc90c04725

ASTERISK-29358 #close

Change-Id: I050daff67066873df4e8fc7f4bd977c1ca06e647

Enhancements:

 * The MessageSend dialplan application now takes an optional
   third argument that can set the message's "To" field on
   outgoing messages.  It's an alternative to using the
   MESSAGE(to) dialplan function.

   NOTE: No channel driver currently implements this field.  A
   follow-on commit for res_pjsip_messaging will implement it for
   the chan_pjsip channel driver.

 * To prevent confusion with the first argument, currently named
   "to", it's been renamed to "destination". Its function,
   creating the request URI, hasn't changed.

 * The documentation for MessageSend was updated to be
   more clear about the parameters and how they interact
   the MESSAGE() dialplan function.

 * With the rename of MessageSend's first parameter, and the fact
   that message.c references <info> elements in chan_sip.c,
   res_pjsip_messaging.c and res_xmpp, they each needed
   documentation updates to use MessageDestinationInfo instead of
   MessageToInfo.

 * appdocsxml.dtd was updated to include a missing element
   declaration for "dataType".  This was showing up as an error
   in Eclipse's dtd editor.

 * Despite the changes in this commit, there should be
   no impact to current users of MessageSend.

Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a

This change embeds the MODULEINFO block of modules
into the core XML documentation. This provides a shared
mechanism for use by both menuselect and Asterisk for
information and a definitive source of truth.

ASTERISK-29335

Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90

The system header strings was included mistakenly with commit 3de02044.
That header is included via asterisk.h and there via the compat.h.

Change-Id: I3dc49060e275295f785670c87cc65fd3c3abd24a

ASTERISK-29105

Change-Id: If1615fe7115fe544ef974b044d3cea5c48b94a38

Instead of looking for pass-through formats in the list of transcodable
formats (which is going to find nothing), go through the result which
is going to be the jointcaps of the tech_pvt of the channel. Finally,
only with that list, ast_format_cap_remove(.) is going to succeed.

This restores the behaviour of Asterisk 1.8. However, it does not fix
ASTERISK_29282 because that issue report is about chan_sip and PJSIP.
Here, only chan_sip is fixed because PJSIP does not even call
ast_rtp_instance_available_formats -> ast_translate_available_format.

Change-Id: Icade2366ac2b82935b95a9981678c987da2e8c34

Two previous commits, 620d9f47 and 6d980de2, allow to set up a call
without audio, again. That was introduced originally with commit f04d5fb8
but changed and broke over time. The original commit missed one
scenario: A [peer] section in sip.conf, which does not allow audio at
all. In that case, chan_sip rejected the call, although even when the
requester offered no audio. Now, chan_sip does not check whether there
is no audio format but checks whether there is no format in general. In
other words, if there is at least one format to offer, the call succeeds.

However, to prevent calls with no-audio, chan_sip still rejects calls
when both call parties (caller = requester of the call *and* callee =
[peer] section in sip.conf) included audio. In such a case, it is
expected that the call should have audio.

ASTERISK-29280

Change-Id: I0fb74faf51ef22a60c10b467df6a4d1c1943b73e

If there's no secret specified for an iax2 peer and there's no secret
specified in the dial string, Asterisk will crash if the auth method
requested by the peer is MD5 or plaintext.  You also couldn't specify
a default auth method in the [general] section of iax.conf so if you
don't have static peers defined and just use the dial string, Asterisk
will still crash even if you have a secret specified in the dial string.

* Added logic to iax2_call() and authenticate_reply() to print
  a warning and hanhup the call if encryption is requested and
  there's no secret or auth method.  This prevents the crash.

* Added the ability to specify a default "auth" in the [general]
  section of iax.conf.

ASTERISK-29624
Reported by: N A

Change-Id: I5928e16137581f7d383fcc7fa04ad96c919e6254

The previous commit 6d980de2 fixed this issue in the core of Asterisk.
With that, each channel technology can be used without audio
theoretically. Practically, the channel-technology driver chan_sip
turned out to have an invalid check preventing that. chan_sip tested
whether there is at least one audio format. However, chan_sip has to
test whether there is at least one format. More cannot be tested while
requesting chan_sip because only the [general] capabilities but not the
[peer] caps are known yet. And the [peer] caps might not be a subset or
show any intersection with the [general] caps. This change here fixes
this.

The original commit f04d5fb8, thirteen years ago, contained a software
bug as it passed ANY audio capability to the channel-technology driver.
Instead, it should have passed NO audio format. Therefore, this
addressed issue here was not noticed in Asterisk 1.6.x and Asterisk 1.8.
Then, Asterisk 10 changed that from ANY to NO, but nobody reported since
then.

ASTERISK-29265

Change-Id: Ic16a3bf13cd1b5c4fc4041ed74961177d96b600f

When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is
0 when no protocl specific error
SIP example of failure, 3xx-6xx for the SIP error code received

This allows applications to perform actions based on the failure
reason.

ASTERISK-29252 #close
Reported-by: Dan Cropp

Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4

This completes the fix for ASTERISK_24543. Only when the call is an
outgoing call, consult and append the configured format capabilities
(p->caps). When all audio formats got rejected the negotiated format
capabilities (p->jointcaps) contain no audio formats for incoming
calls. This is required when there are other accepted media streams.

ASTERISK-29258

Change-Id: I8bab31c7f3f3700dce204b429ad238a524efebb9

There are a couple of parameters (datalen and data) that do not get set
in chan_pjsip_indicate which could cause an Invalid message to pop up
for things such as fax. This patch adds them to the frame.

Change-Id: Ia51be086a0708be905e73d1f433572c49c7e38f8

session->channel doesn't exist until chan_pjsip creates it, so intead of
setting a channel variable every new incoming call sets one and the same
global variable.

This patch moves the code to chan_pjsip so that SIPDOMAIN is set on
a newly created channel, it also removes a misleading reference to
channel->session used to fetch call pickup configuraion.

ASTERISK-29240

Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755

Previously, chan_sip parsed all known media streams in an SDP offer
like video (and text) even when videosupport=no (and textsupport=no).
This wasted processor power. Furthermore, chan_sip accepted SDP offers,
including no audio but just video (or text) streams although
videosupport=no (or textsupport=no). Finally, chan_sip denied the whole
offer instead of individual streams when they had encryption (SDES-sRTP)
unexpectedly enabled.

ASTERISK-29238
ASTERISK-29237
ASTERISK-29222

Change-Id: Ie49e4e2a11f0265f914b684738348ba8c0f89755

The fix for ASTERISK-27902 made chan_pjsip process SIP responses twice.
This resulted in extra noise in logs (for example, "is making progress"
and "is ringing" get logged twice by app_dial), as well as in noise in
signalling: one incoming 183 Session Progress results in 2 outgoing 183-s.

This change splits the response handler into 2 functions:
 - one for updating HANGUPCAUSE, which is still called twice,
 - another that does the rest, which is called only once as before.

ASTERISK-28016
Reported-by: Alex Hermann

ASTERISK-28549
Reported-by: Gant Liu

ASTERISK-28185
Reported-by: Julien

Change-Id: I0a1874be5bb5ed12d572d17c7f80de6e5e542940

Add channel reference count for PJSIP REFER. The call could be terminated
prior to the result of the transfer. In that scenario, when the SUBSCRIBE/NOTIFY
occurred several minutes later, it would attempt to access a session which was
no longer valid.  Terminate event subscription if pjsip_xfer_initiate() or
pjsip_xfer_send_request() fails in transfer_refer().

ASTERISK-29201 #close
Reported-by: Dan Cropp

Change-Id: I3fd92fd14b4e3844d3d7b0f60fe417a4df5f2435

Fix the usual compile problem when someone adds a new callback to struct
ast_channel_tech.

Change-Id: I9bdeb8a8cc65f03b2d6e4f2eb5809af47c906c32

In some circumstances it was possible for an INVITE
session to be destroyed while we were still using it.
This occurred due to the reference on the INVITE session
being released internally as a result of its state
changing to DISCONNECTED.

This change adds a reference to the INVITE session
which is released when our own session is destroyed,
ensuring that the INVITE session remains valid for
the lifetime of our session.

ASTERISK-29022

Change-Id: I300c6d9005ff0e6efbe1132daefc7e47ca6228c9

12 years ago, with ASTERISK_12115 the last four get/uses of socket.port
vanished. However, the struct member itself and all seven set/uses
remained as dead code.

ASTERISK-28798

Change-Id: Ib90516a49eca3d724a70191278aaf2144fb58c59

ASTERISK-29144

Change-Id: I2a72c072083b4492a223c6f9d73d21f4f424db62