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

The DBGetTree AMI action's ListItem previously
always reported 1, regardless of the count. This
is corrected to report the actual count.

ASTERISK-30245 #close
patches:
  gettreecount.diff submitted by Birger Harzenetter (license 5870)

Change-Id: I46d8992710f1b8524426b1255f57d1ef4a4934d4

The IF function currently emits warnings if both IF branches
are empty. However, there is no actual necessity that either
branch be non-empty as, unlike other conditional applications/
functions, nothing is inherently done with IF, and both
sides could legitimately be empty. The warning is thus turned
into a debug message.

ASTERISK-30243 #close

Change-Id: I5250625dd720f95e1859b5dfb933905d7e7a730e

Adds the n "no answer" option to the Bridge application
so that answer supervision can not automatically
be provided when Bridge is executed.

Additionally, a mechanism (dialplan variable)
is added to prevent bridge targets (typically the
target of a masquerade) from answering the channel
when they enter the bridge.

ASTERISK-30223 #close

Change-Id: I76f73fcd8e403bcd18f2abb40c658f537ac1ba6d

Adds the n option to not answer the channel when calling
BridgeWait, so the application can be used without
forcing answer supervision.

ASTERISK-30216 #close

Change-Id: I6b85ef300b1f7b5170f8537e2b10889cc2e6605a

Adds an option that will play an audio file
to the party while AMD is running on the
channel, so the called party does not just
hear silence.

ASTERISK-30179 #close

Change-Id: I4af306274552b61b3d9f0883c33f698abd4699b6

ASTERISK-30232 #close

Change-Id: I2603e2cef8f93f6b0a6ef39f7eac744251bb3902

Adds the EXPORT function, which allows write
access to variables and functions on other
channels.

ASTERISK-29432 #close

Change-Id: I7492645ae4307553d0f586d78e13a4f586231fdf

This patch adds a new option to the 100rel parameter for pjsip
endpoints called "peer_supported". When an endpoint with this option
receives an incoming request and the request indicated support for the
100rel extension, then Asterisk will send 1xx responses reliably. If
the request did not indicate 100rel support, Asterisk sends 1xx
responses normally.

ASTERISK-30158

Change-Id: Id6d95ffa8f00dab118e0b386146e99f254f287ad

Fix segfault due to null pointer dereference
inside the audiohook callback.

ASTERISK-30220 #close

Change-Id: Ideb80f606974366e89d619d908744230b5a5a259

If we find that n_max (currently hard wired to 1) sessions were purged,
schedule the next purge for 1ms into the future rather than 5000ms (as
per current).  This way we will purge up to 1000 sessions per second
rather than 1 every 5 seconds.

This mitigates a build-up of sessions should http sessions gets
established faster than 1 per 5 seconds.

Change-Id: I9820d39aa080109df44fe98c1325cafae48d54f5
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

Adds TRIM, LTRIM, and RTRIM, which can be used
for trimming leading and trailing whitespace
from strings.

ASTERISK-30222 #close

Change-Id: I50fb0c40726d044a7a41939fa9026f3da4872554

ASTERISK-30235

Change-Id: Ia1e326e7b52cd06fd5e6c9009e3e63193c92f6cd

Fixed "may be used uninitialized" errors in geoloc_config.c.

ASTERISK-30234

Change-Id: I1ea336bf7abbc16fa59b75720f0db8f1d960b3d4

ASTERISK-30226 #close

Change-Id: I5695fb0c9521f112f754b8362cff2a8f3eff05c5