Currently, there is no Caller ID available to us when
checking for an extension match when handling INVITEs.
As a result, extension patterns that depend on the Caller ID
are not matched and calls may be incorrectly rejected.

The Caller ID is not available because the supplement that
adds Caller ID to the session does not execute until after
this check. Supplement callbacks cannot yet be executed
at this point since the session is not yet in the appropriate
state.

To fix this without impacting existing behavior, the Caller ID
number is now retrieved before attempting to pattern match.
This ensures pattern matching works correctly and there is
no behavior change to the way supplements are called.

ASTERISK-28767 #close

Change-Id: Iec7f5a3b90e51b65ccf74342f96bf80314b7cfc7

When a call is put on hold and it has moh_passthrough and rtp_timeout
set on the endpoint, the wrong timeout will be used. rtp_timeout_hold is
expected to be used, but rtp_timeout is used instead. This change adds a
couple of checks for locally_held to determine if rtp_timeout_hold needs
to be used instead of rtp_timeout.

ASTERISK-30350

Change-Id: I7b106fc244332014216d12bba851cefe884cc25f

Fixes a negative offset warning by initializing
the buffer to empty.

Additionally, although it doesn't currently complain
about it, the size of a buffer is increased to
accomodate the maximum size contents it could have.

ASTERISK-30240 #close

Change-Id: I8eecedf14d3f2a75864797f802277cac89a32877

Currently, if a user attempts to set a Caller ID related
function to an invalid value, a warning is emitted,
except for when setting the redirecting reason.
We now emit a warning if we were unable to successfully
parse the user-provided reason.

ASTERISK-30332 #close

Change-Id: Ic341f5d5f7303b6f1115549be64db58a85944f5a

Fix aor lookup on sip path addition. Issue happens in case of dialing
with @ and overriding user part of RURI.

ASTERISK-30100 #close
Reported-by: Yury Kirsanov

Change-Id: I3f2c42a583578c94397b113e32ca3ebf2d600e13

When ast_stream_and_wait returns an error (for example, when attempting
to stream to a channel after hangup) the stream is not closed, and
callers typically do not check the return code. This results in leaking
file descriptors, leading to resource exhaustion.

This change ensures that the stream is closed in case of error.

ASTERISK-30198 #close
Reported-by: Julien Alie

Change-Id: Ie46b67314590ad75154595a3d34d461060b2e803

Removes see-also references to applications that don't
exist anymore (removed in Asterisk 19),
so these dead links don't show up on the wiki.

ASTERISK-30347 #close

Change-Id: I9539bc30f57cd65aa4e2d5ce8185eafa09567909

The `ast_geoloc_datastore_add_eprofile` function does not return 0 on
success, it returns the size of the underlying datastore. This means
that the datastore will be freed and its pointer set to NULL when no
error occured at all.

ASTERISK-30346

Change-Id: Iea9b209bd1244cc57b903b9496cb680c356e4bb9

When adding AOC to an outgoing response the code
assumed that a body would exist for comparing the
Content-Type. This isn't always true.

The code now checks to make sure the response has
a body before checking the Content-Type.

ASTERISK-21502

Change-Id: Iaead371434fc3bc693dad487228106a7d7a5ac76

Fixes format truncation warnings in gcc 12.2.1.

ASTERISK-30349 #close

Change-Id: I42be4edf0284358b906e765d1966b6b9d66e1d3c

ASTERISK-21502

Change-Id: I051b778f8c862d3b4794d28f2f3d782316707b08

chan_sip supported sending AOC-D and AOC-E information in SIP INFO
messages in an "AOC" header in a format that was originally defined by
Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC
format that is supported by devices from multiple vendors, including
Snom phones with firmware >= 8.4.2 (released in 2010).

This commit adds a new res_pjsip_aoc module that inserts AOC information
into outgoing messages or sends SIP INFO messages as described below.
It also fixes a small issue in res_pjsip_session which didn't always
call session supplements on outgoing_response.

* AOC-S in the 180/183/200 responses to an INVITE request
* AOC-S in SIP INFO (if a 200 response has already been sent or if the
  INVITE was sent by Asterisk)
* AOC-D in SIP INFO
* AOC-D in the 200 response to a BYE request (if the client hangs up)
* AOC-D in a BYE request (if Asterisk hangs up)
* AOC-E in the 200 response to a BYE request (if the client hangs up)
* AOC-E in a BYE request (if Asterisk hangs up)

The specification defines one more, AOC-S in an INVITE request, which
is not implemented here because it is not currently possible in
Asterisk to have AOC data ready at this point in call setup. Once
specifying AOC-S via the dialplan or passing it through from another
SIP channel's INVITE is possible, that might be added.

The SIP INFO requests are sent out immediately when the AOC indication
is received. The others are inserted into an appropriate outgoing
message whenever that is ready to be sent. In the latter case, the XML
is stored in a channel variable at the time the AOC indication is
received. Depending on where the AOC indications are coming from (e.g.
PRI or AMI), it may not always be possible to guarantee that the AOC-E
is available in time for the BYE.

Successfully tested AOC-D and both variants of AOC-E with a Snom D735
running firmware 10.1.127.10. It does not appear to properly support
AOC-S however, so that could only be tested by inspecting SIP traces.

ASTERISK-21502 #close
Reported-by: Matt Jordan <mjordan@digium.com>

Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333

When passing a JSON body to the 'create' channel route
it would be converted into Asterisk variables, but never
freed resulting in a memory leak.

This change makes it so that the variables are freed in
all cases.

ASTERISK-30344

Change-Id: I924dbd866a01c6073e2d6fb846ccaa27ef72d49d

msg_create_from_file currently does not dispatch emails,
which means that applications using this function, such
as MixMonitor, will not trigger notifications to users
(only AMI events are sent our currently). This is inconsistent
with other ways users can receive voicemail.

This is fixed by adding an option that attempts to send
an email and falling back to just the notifications as
done now if that fails. The existing behavior remains
the default.

ASTERISK-30283 #close

Change-Id: I597cbb9cf971a18d8776172b26ab187dc096a5c7

This fixes a small typo in the from_domain documentation on the endpoint documentation

ASTERISK-30328 #close

Change-Id: Ia6f0897c3f5cab899ef2cde6b3ac07265b8beb21

Adds support for the capture agent name field
of the Homer protocol to Asterisk by allowing
users to specify a name that will be sent to
the HEP server.

ASTERISK-30322 #close

Change-Id: I6136583017f9dd08daeb8be02f60fb8df4639a2b

Adds the If, ElseIf, Else, ExitIf, and EndIf
applications for conditional execution
of a block of dialplan, similar to the While,
EndWhile, and ExitWhile applications. The
appropriate branch is executed at most once
if available and may be broken out of while
inside.

ASTERISK-29497

Change-Id: I3aa3bd35a5add82465c6ee9bd86b64601f0e1f49

Some SIP devices use an empty extension for PLAR functionality.

Rather than rejecting these empty extensions, we now use the s
extension for such calls to mirror the existing PLAR functionality
in Asterisk (e.g. chan_dahdi).

ASTERISK-30265 #close

Change-Id: I0861a405cd49bbbf532b52f7b47f0e2810832590

Updates the documentation for the 'contact_user' field to point out the
default outbound contact if no contact_user is specified 's'

ASTERISK-30316 #close

Change-Id: I61f24fb9164e4d07e05908a2511805281874c876

The commit that rearchitected media formats,
a2c912e9 (ASTERISK_23114)
introduced a regression by improperly translating code in res_adsi.c.
In particular, the pointer to the frame buffer was initialized
at the top of adsi_careful_send, rather than dynamically updating it
for each frame, as is required.

This resulted in the first frame being repeatedly sent,
rather than advancing through the frames.
This corrupted the transmission of the CAS to the CPE,
which meant that CPE would never respond with the DTMF acknowledgment,
effectively completely breaking ADSI functionality.

This issue is now fixed, and ADSI now works properly again.

ASTERISK-29793 #close

Change-Id: Icdeddf733eda2981c98712d1ac9cddc0db507dbe

Adds support for custom URI and header parameters
in the From header in PJSIP. Parameters can be
both set and read using this function.

ASTERISK-30150 #close

Change-Id: Ifb1bc3c512ad5f6faeaebd7817f004a2ecbd6428

When parsing information from AstDB while loading,
it is possible that certain pointers are never
set, which leads to invalid memory access and
then, fatally, invalid free attempts on this memory.
We now initialize to NULL to prevent this.

ASTERISK-30311 #close

Change-Id: I6120681d04fd2c12a9473f35ce95a1f8e74e3929

ASTERISK_28702 previously attempted to fix an
issue with flash hook hold timing out after
just under 17 minutes, when it should have never
been timing out. It fixed this by changing 999999
to INT_MAX, but it did so in chan_dahdi, which
is the wrong place since ss_thread is now in
sig_analog and the one in chan_dahdi is mostly
dead code.

This fixes this by porting the fix to sig_analog.

ASTERISK-30336 #close

Change-Id: I05eb69cc0b5319d357842a70bd26ef64d145cb15

The XML docs are currently only loaded on
startup with no way to update them during runtime.
This makes it impossible to load modules that
use ACO/Sorcery (which require documentation)
if they are added to the source tree and built while
Asterisk is running (e.g. external modules).

This adds a CLI command to reload the XML docs
during runtime so that documentation can be updated
without a full restart of Asterisk.

ASTERISK-30289 #close

Change-Id: I4f265b0e5517e757c5453a0f241201a5788d3a07

This file includes some doxygen comments referencing
ast_format_set. This is an obsolete API that was
removed years back, but documentation was not fully
updated to reflect that. These examples are
updated to the current way of doing things
(using the format cache).

ASTERISK-30327 #close

Change-Id: I570f3b8007fa17ba470cc7117f44bfe7c555d2f7

MixMonitor currently uses the Connected Line as the Caller ID
for voicemails. This is due to the implementation being written
this way for use with Digium phones. However, in general this
is not correct for generic usage in the dialplan, and people
may need the real Caller ID instead. This adds an option to do that.

ASTERISK-30286 #close

Change-Id: I3d0ce76dfe75e2a614e0f709ab27acbd2478267c

Backports two security fixes (c4d3498 and 450baca) from pjproject 2.13.

ASTERISK-30338

Change-Id: I86fdc003d5d22cb66e7cc6dc3313a8194f27eb69

It was possible for a module that registered for transport monitor
events to pass in a pjsip_transport that had already been freed.
This caused pjsip_transport_events to crash when looking up the
monitor for the transport.  The fix is a two pronged approach.

1. We now increment the reference count on pjsip_transports when we
create monitors for them, then decrement the count when the
transport is going to be destroyed.

2. There are now APIs to register and unregister monitor callbacks
by "transport key" which is a string concatenation of the remote ip
address and port.  This way the module needing to monitor the
transport doesn't have to hold on to the transport object itself to
unregister.  It just has to save the transport_key.

* Added the pjsip_transport reference increment and decrement.

* Changed the internal transport monitor container key from the
  transport->obj_name (which may not be unique anyway) to the
  transport_key.

* Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that
  fills a buffer with the transport_key using a passed-in
  pjsip_transport.

* Added the following functions:
  ast_sip_transport_monitor_register_key
  ast_sip_transport_monitor_register_replace_key
  ast_sip_transport_monitor_unregister_key
  and marked their non-key counterparts as deprecated.

* Updated res_pjsip_pubsub and res_pjsip_outbound_register to use
  the new "key" monitor functions.

NOTE: res_pjsip_registrar also uses the transport monitor
functionality but doesn't have a persistent object other than
contact to store a transport key.  At this time, it continues to
use the non-key monitor functions.

ASTERISK-30244

Change-Id: I1a20baf2a8643c272dcf819871d6c395f148f00b

Add live_dangerously flag to manager and use this flag to
determine if a configuation file outside of AST_CONFIG_DIR
should be read.

ASTERISK-30176

Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404

When decoding q.931 encoded calling/called number
now checking for length being less than minimum required.

ASTERISK-30103

Change-Id: I3dcfce0f35eca258dc450f87c92d4d7af402c2e7

The Answer application currently waits for up to 500ms
for media, even if users specify a different timeout.

This adds an option to not wait for media on the channel
by doing a raw answer instead. The default 500ms threshold
is also documented.

ASTERISK-30308 #close

Change-Id: Id59cd340c44b8b8b2384c479e17e5123e917cba4

Currently, chan_dahdi will wait for at least one
ring before an incoming call can enter the dialplan.
This is generally necessary in order to receive
the Caller ID spill and/or distinctive ringing
detection.

However, if neither of these is required, then there
is nothing gained by waiting for one ring and this
unnecessarily delays call setup. Users can now
use immediate=yes to make FXO channels (FXS signaled)
begin processing dialplan as soon as Asterisk receives
the call.

ASTERISK-30305 #close

Change-Id: I20818b370b2e4892c7f40c8a8753fa06a81750b5

This PR contains two relatively separate changes in channel.c and
res_pjsip_session.c which ensure that topology changes are not ignored
in cases where they should be handled.

For channel.c:

The function ast_channel_request_stream_topology_change only triggers a
stream topology request change indication, if the channel's topology
does not equal the requested topology. However, a channel could be in a
state where it is currently "negotiating" a new topology but hasn't
updated it yet, so the topology request change would be lost. Channels
need to be able to handle such situations internally and stream
topology requests should therefore always be passed on.

In the case of chan_pjsip for example, it queues a session refresh
(re-INVITE) if it is currently in the middle of a transaction or has
pending requests (among other reasons).

Now, ast_channel_request_stream_topology_change always indicates a
stream topology request change even if the requested topology equals the
channel's topology.

For res_pjsip_session.c:

The function resolve_refresh_media_states does not process stream state
changes if the delayed active state differs from the current active
state. I.e. if the currently active stream state has changed between the
time the sip session refresh request was queued and the time it is being
processed, the session refresh is ignored. However, res_pjsip_session
contains logic that ensures that session refreshes are queued and
re-queued correctly if a session refresh is currently not possible. So
this check is not necessary and led to some session refreshes being
lost.

Now, a session refresh is done even if the delayed active state differs
from the current active state and it is checked whether the delayed
pending state differs from the current active - because that means a
refresh is necessary.

Further, the unit test of resolve_refresh_media_states was adapted to
reflect the new behavior. I.e. the changes to delayed pending are
prioritized over the changes to current active because we want to
preserve the original intention of the pending state.

ASTERISK-30184

Change-Id: Icd0703295271089057717006730b555b9a1d4e5a

SLAStation currently autoservices the station channel before
creating a thread to actually dial the trunk. This leads
to duplicate servicing of the channel which causes assertions,
deadlocks, crashes, and moreover not the correct behavior.

Removing the autoservice prevents the crash, but if the station
hangs up before the trunk answers, the call hangs since the hangup
was never serviced on the channel.

This is fixed by not autoservicing the channel, but instead
servicing it in the thread dialing the trunk, since it is doing
so synchronously to begin with. Instead of sleeping for 100ms
in a loop, we simply use the channel for timing, and abort
if it disappears.

The same issue also occurs with SLATrunk when a call is answered,
because ast_answer invokes ast_waitfor_nandfds. Thus, we use
ast_raw_answer instead which does not cause any conflict and allows
the call to be answered normally without thread blocking issues.

ASTERISK-29998 #close

Change-Id: Icc237d50354b5910000d2305901e86d2c87bb9d8

Found in res_geolocation, but I believe others may have similar issues,
thus not linking to a specific issue.

Essentially gcc doesn't mark the stack for being non-executable unless
it's compiling the source, this informs ld via gcc to mark the object as
not requiring an executable stack (which a binary blob obviously
doesn't).

ASTERISK-30321

Change-Id: I71bcc2fd1fe0c82a28b3257405d6f2b566fd9bfc
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

A memory leak was present in func_json due to
using ast_json_free, which just calls ast_free,
as opposed to recursively freeing the JSON
object as needed. This is now fixed to use the
right free functions.

ASTERISK-30293 #close

Change-Id: I982324dde841dc9147c8d8ad35c8719daf418b49

Removes the function mkstemp_file and uses
ast_file_mkftemp from file.h instead.

ASTERISK-30295 #close

Change-Id: I7412ec06f88c39ee353bcdb8c976c2fcac546609

The "RECORD FILE" command in res_agi has its own
implementation for actually doing the recording. This
has resulted in it not actually obeying the option
"transmit_silence" when recording.

This change causes it to now send silence if the
option is enabled.

ASTERISK-30314

Change-Id: Ib3a85601ff35d1b904f836691bad8a4b7e957174

Adds an option that allows MixMonitor to delete
its copy of any recording files before exiting.

This can be handy in conjunction with options
like m, which copy the file elsewhere, and the
original files may no longer be needed.

ASTERISK-30284 #close

Change-Id: Ida093679c67e300efc154a97b6d8ec0f104e581e