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

The ModuleCheck XML documentation falsely
claims that the module's version number is returned.
This has not been the case since 14, since the version
number is not available anymore, but the documentation
was not changed at the time. It is now updated to
reflect this.

ASTERISK-30285 #close

Change-Id: Idde2d1205a11f2623fa1ddab192faa3dc4081e91

Adds an ignore case for wink since it should
pass through with no warning.

ASTERISK-30290 #close

Change-Id: Ieb7e34daa717357ac5c93efb0059f6c2321f16ad

Fixed the specification of "outputdir" when calling ast_coredumper
so the txt files are saved in the correct place.

ASTERISK-30282

Change-Id: Ic631cb90c1e4c29d970c982dff45fda5e0eb15b6

When gosub is executed on channels without a PBX, the context,
extension, and priority are initialized to the channel driver's
default location for that endpoint. As a result, the last Return
will restore this location and the Gosub logs will print out bogus
information about our exit point.

To fix this, on channels that don't have a PBX, the execution
location is left intact on the last return if there are no
further stack frames left. This allows the correct location
to be printed out to the user, rather than the bogus default
context.

ASTERISK-30076 #close

Change-Id: I1d42a99c9aa9e3708d32718863175158a894e414

unicast_rtp_request() was setting the channel variables like this:

pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS",
    ast_sockaddr_stringify_addr(&local_address));
ast_rtp_instance_get_local_address(instance, &local_address);
pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT",
    ast_sockaddr_stringify_port(&local_address));

...which made it appear that UNICASTRTP_LOCAL_ADDRESS was being
set before local_address was set.  In fact, the address part of
local_address was set earlier in the function, just not the port.
This was confusing however so ast_rtp_instance_get_local_address()
is now being called before setting UNICASTRTP_LOCAL_ADDRESS.

ASTERISK-30281

Change-Id: I872ac49477100f4eb33891d46efc6ca21ec81aa4

When a websocket (or potentially any stateful connection) is quickly
created then destroyed, it is possible that the qualify thread will
destroy the transaction before the initialzing thread is finished
with it.

Depending on the timing, this can cause an assertion within pjsip.

To prevent this, ast_send_stateful_response will now create the group
lock and add a reference to it before creating the transaction.

While this should resolve the crash, there is still the potential that
the contact will not be cleaned up properly, see:ASTERISK~29286. As a
result, the contact has to 'time out' before it will be removed.

ASTERISK-28689

Change-Id: Id050fded2247a04d8f0fc5b8a2cf3e5482cb8cee

write_openssl_error_to_log has been erroneously
using ast_free instead of free, which will
cause a crash when MALLOC_DEBUG is enabled since
the memory was not allocated by Asterisk's memory
manager. This changes it to use the actual free
function directly to avoid this.

ASTERISK-30278 #close

Change-Id: Iac8b6468b718075809c45d8ad16b101af21a474d

Current registration code use pjsip_parse_uri to verify outbound_proxy
that is different from the reading this option for the endpoint. This
made value with multiple proxies invalid for registration pjsip settings.
Removing URI validation helps to use registration through multiple proxies.

ASTERISK-30217 #close

Change-Id: I064558e66f04b9f3260c46181812a01349761357

Fix compilation errors caused by using size_t
instead of uintmax_t and non-portable format
specifiers.

ASTERISK-30273 #close

Change-Id: I363e6057ef84d54b88af80d23ad6147eef9216ee

Currently chan_pjsip on receiving a re-INVITE without SDP will only
return the codecs that are previously negotiated and not offering
all enabled codecs.

This causes interoperability issues with different equipment (e.g.
from Cisco) for some of our customers and probably also in other
scenarios involving 3PCC infrastructure.

According to RFC 3261, section 14.2 we SHOULD return all codecs
on a re-INVITE without SDP

The PR proposes a new parameter to configure this behaviour:
all_codecs_on_empty_reinvite. It includes the code, documentation,
alembic migrations, CHANGES file and example configuration additions.

ASTERISK-30193 #close

Change-Id: I69763708d5039d512f391e296ee8a4d43a1e2148

The PJSIP notify CLI commands allow for using
"options" configured in pjsip_notify.conf.

This allows these same options to be used in
AMI actions as well.

Additionally, as part of this improvement,
some repetitive common code is refactored.

ASTERISK-30263 #close

Change-Id: Ie4496b322b63b61eaf9672183a959ab99a04b6b5

Expands the pjsip logger to support the ability to filter
by SIP message method. This can make certain types of SIP debugging
easier by only logging messages of particular method(s).

ASTERISK-30146 #close

Co-authored-by: Sean Bright <sean@seanbright.com>
Change-Id: I9c8cbb6fc8686ef21190eb42e08bc9a9b147707f

race condition: ast_dial_join() may not cancel outgoing call, if
function is called just after called party answer and before
application execution (bit is_running_app not yet set).

This fix adds ast_softhangup() calls in addition to existing
pthread_kill() when is_running_app is not set.

ASTERISK-30258

Change-Id: Idbdd5c15122159661aa8e996a42d5800083131e4

This fixes dahdi_request to properly set the cause
code to CONGESTION instead of BUSY if no channels
were actually available.

Currently, the cause is erroneously set to busy
if the channel itself is found, regardless of its
current state. However, if the channel is not available
(e.g. T1 down, card not operable, etc.), then the
channel itself may not be in a functional state,
in which case CHANUNAVAIL is the correct cause to use.

This adds a simple check to ensure that busy tone
is only returned if a channel is encountered that
has an owner, since that is the only possible way
that a channel could actually be busy.

ASTERISK-30274 #close

Change-Id: Iad5870223c081240c925b19df8d6af136953b994

pjproject does not provide any mechanism of removing
event packages, which means that once a subscription
handler is registered, it is effectively permanent.

pjproject will assert if the same event package is
ever registered again, so currently unloading and
loading any Asterisk modules that use subscriptions
will cause a crash that is beyond our control.

For that reason, we now prevent users from being
able to unload these modules, to prevent them
from ever being loaded twice.

ASTERISK-30264 #close

Change-Id: I7fdcb1a5e44d38b7ba10c44259fe98f0ae9bc12c

Some logic in say.c for determining if we need
to also add an ampersand for file seperation was faulty,
as non-successful files would increment the count, causing
a leading ampersand to be added improperly.

This is fixed, and a unit test that captures this regression
is also added.

ASTERISK-30248 #close

Change-Id: I02c1d3a11d82fe4ea8b462070cbd1effb5834d2b

ASTERISK-30213 #close

Change-Id: I4a77143d41615b7c4fc25bb1251c0a9cb87b417a

Add enum to allow setting optional direction. If set to only one
direction, only feed matching-direction frames to the associated
slin factory.

This prevents mangling the transcoder on non-mixed frames when the
READ and WRITE frames would have otherwise required it.  Also
removes the need to mute or discard the un-wanted frames as they
are no longer added in the first place.

res_stasis_snoop is changed to use this addition to set direction
on audiohook based on spy direction.

If no direction is set, the ast_audiohook_init will init this enum
to BOTH which maintains existing functionality.

ASTERISK-30252

Change-Id: If8716bad334562a5d812be4eeb2a92e4f3be28eb

Allows bridging, parking, and dial messages to be globally
ignored for all CDRs such that only a single CDR record
is generated per channel.

This is useful when CDRs should endure for the lifetime of
an entire channel and bridging and dial updates in the
dialplan should not result in multiple CDR records being
created for the call. With the ignore bridging option,
bridging changes have no impact on the channel's CDRs.
With the ignore dial state option, multiple Dials and their
outcomes have no impact on the channel's CDRs. The
last disposition on the channel is preserved in the CDR,
so the actual disposition of the call remains available.

These two options can reduce the amount of "CDR hacks" that
have hitherto been necessary to ensure that CDR was not
"spoiled" by these messages if that was undesired, such as
putting a dummy optimization-disabled local channel between
the caller and the actual call and putting the CDR on the channel
in the middle to ensure that CDR would persist for the entire
call and properly record start, answer, and end times.
Enabling these options is desirable when calls correspond
to the entire lifetime of channels and the CDR should
reflect that.

Current default behavior remains unchanged.

ASTERISK-30091 #close

Change-Id: I393981af42732ec5ac3ff9266444abb453b7c832

Adds support for detecting audible ringback tone
to the TONE_DETECT function using the p option.

ASTERISK-30254 #close

Change-Id: Ie2329ff245248768367d26749c285fbe823f6414

Fixes a format truncation warning in notify_message.

ASTERISK-30256 #close

Change-Id: I983a423c0214641ca4f8c9dfe0b19c47448fdee1

"fname" is passed in as a const char *, but strstr() mangles that
into a char *, and we were attempting to modify the string in place.
This is an unwanted (and undocumented) side-effect.

ASTERISK-30213

Change-Id: Ifa36d352aafeb7f9beec3f746332865c7d21e629

ASTERISK-30213

Change-Id: I115f5f8942ffcfb23cd2559a55bac8a2eba081e0

Also added a note to the geolocation.conf.sample file
and added a README to the res/res_geolocation/wiki
directory.

Change-Id: I89c3c5db8c0701b33127993622d5e4f904bddfbc

This patch adds support for mediasec SIP headers and SDP attributes.
These are defined in RFC 3329, 3GPP TS 24.229 and
draft-dawes-sipcore-mediasec-parameter. The new features are
implemented so that a backbone for RFC 3329 is present to streamline
future work on RFC 3329.

With this patch, Asterisk can communicate with Deutsche Telekom trunks
which require these fields.

ASTERISK-30032

Change-Id: Ia7f5b5ba42db18074fdd5428c4e1838728586be2

Avoid crashing by skipping invisible bridges and checking the
snapshot for a null pointer. In effect this is how the bridges
are enumerated in res/ari/resource_bridges.c already.

ASTERISK-30239
ASTERISK-30237

Change-Id: I58ef9f44036feded5966b5fc70ae754f8182883d

If geolocation is not in use for an endpoint, the NOTICE
log level is currently spammed with messages about this,
even though nothing is wrong and these messages provide
no real value. These log messages are therefore changed
to debugs.

ASTERISK-30241 #close

Change-Id: I656b355d812f67cc0f0fdf09b00b0e1458598bb4