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

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

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

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

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

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

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

ASTERISK-30254 #close

Change-Id: Ie2329ff245248768367d26749c285fbe823f6414

"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

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

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

ASTERISK-30234

Change-Id: I1ea336bf7abbc16fa59b75720f0db8f1d960b3d4

ASTERISK-30226 #close

Change-Id: I5695fb0c9521f112f754b8362cff2a8f3eff05c5

Adding user=phone to local-side uri's when user_eq_phone=yes is set for
an endpoint. Previously this would only add the header to the To and R-URI.

ASTERISK-30178

Change-Id: Id3bfb5d225d762e7d2668c023fe09e4541ae8600

Fixed a segfault caused by var_list_from_loc_info() encountering
an empty location info element.

Fixed an issue in ast_strsep() where a value with only whitespace
wasn't being preserved.

Fixed an issue in ast_variable_list_from_quoted_string() where
an empty value was considered a failure.

ASTERISK-30215
Reported by: Dan Cropp

Change-Id: Ieca64e061a6d9298f0196c694b60d986ef82613a

This change adds an option, answeredonly, that will prevent music on
hold on channels that are not answered.

ASTERISK-30135

Change-Id: I1ab0defa43a29a26ae39f94c623596cf90fddc08

This change allows TEL URI requests to come through for basic calls. The
allowed requests are INVITE, ACK, BYE, and CANCEL. The From and To
headers will now allow TEL URIs, as well as the request URI.

Support is only for TEL URIs present in traffic from a remote party.
Asterisk does not generate any TEL URIs on its own.

ASTERISK-26894

Change-Id: If5729e6cd583be7acf666373bf9f1b9d653ec29a

ASTERISK-30046 #close

Change-Id: I5c738756de75fd27ebad54be144c0ac6193f21b2

We're validating the following functionality:

encrypting a block of data with RSA
decrypting a block of data with RSA
signing a block of data with RSA
verifying a signature with RSA
encrypting a block of data with AES-ECB
encrypting a block of data with AES-ECB

as well as accessing test keys from the keystore.

ASTERISK-30045 #close

Change-Id: I0d10e7b41009c5290a4356c6480e636712d5c96d

ASTERISK-30045

Change-Id: If59bbb50c1771084bfe2fef307a6077c90d35ce8

ASTERISK-30046

Change-Id: Ie77e0648f8b0b1c2159fb24662d1989cfd4cc36d

Added an 'a' option to the GEOLOC_PROFILE function to allow
variable lists like location_info_refinement to be appended
to instead of replacing the entire list.

Added an 'r' option to the GEOLOC_PROFILE function to resolve all
variables before a read operation and after a Set operation.

Added a few missing parameters to the ones allowed for writing
with GEOLOC_PROFILE.

Fixed a bug where calling GEOLOC_PROFILE to read a parameter
might actually update the profile object.

Cleaned up XML documentation a bit.

ASTERISK-30190

Change-Id: I75f541db43345509a2e86225bfa4cf8e242e5b6c

You can now specify the location object's format, location_info,
method, location_source and confidence parameters directly on
a profile object for simple scenarios where the location
information isn't common with any other profiles.  This is
mutually exclusive with setting location_reference on the
profile.

Updated appdocsxml.dtd to allow xi:include in a configObject
element.  This makes it easier to link to complete configOptions
in another object.  This is used to add the above fields to the
profile object without having to maintain the option descriptions
in two places.

ASTERISK-30185

Change-Id: Ifd5f05be0a76f0a6ad49fa28d17c394027677569

Added profile parameter "suppress_empty_ca_elements" that
will cause Civic Address elements that are empty to be
suppressed from the outgoing PIDF-LO document.

Fixed a possible SEGV if a sub-parameter value didn't have a
value.

ASTERISK-30177

Change-Id: I924ccc5aa2f45110a3155b22e53dfaf3ef2092dd

The trigger to perform outgoing geolocation processing is the
presence of a geoloc_outgoing_call_profile on an endpoint. This
is intentional so as to not leak location information to
destinations that shouldn't receive it.   In a totally dynamic
configuration scenario however, there may not be any profiles
defined in geolocation.conf.  This makes it impossible to do
outgoing processing without defining a "dummy" profile in the
config file.

This commit adds 4 built-in profiles:
  "<prefer_config>"
  "<discard_config>"
  "<prefer_incoming>"
  "<discard_incoming>"
The profiles are empty except for having their precedence
set and can be set on an endpoint to allow processing without
entries in geolocation.conf.  "<discard_config>" is actually the
best one to use in this situation.

ASTERISK-30182

Change-Id: I1819ccfa404ce59802a3a07ad1cabed60fb9480a

When producing an outgoing SDP we iterate through the configured
formats and produce SDP information. It is possible for some
configured formats to not have SDP information available. If this
is the case we skip over them to allow the SDP to still be
produced.

ASTERISK-29185

Change-Id: I3e37569aa4ca341260e6ca5904dc2f75e46a1749

This change adds support using the pjsip_tls_transport_restart
function for reloading the TLS certificate and key, if the filenames
remain unchanged. This is useful for Let's Encrypt and other
situations. Note that no restart of the transport will occur if
the certificate and key remain unchanged.

ASTERISK-30186

Change-Id: I9bc95a6bf791830a9491ad9fa43c17d4010028d0

Fixes two typos that cause fax detection to not work.
One refers to the wrong frame variable, and the other
refers to the subclass.integer instead of the frametype
as it should.

ASTERISK-30192 #close

Change-Id: I7b35fdb7bcf25a29a212eee37c20812c64ab3ef1

Set termination state to old subscriptions to prevent queueing and sending
NOTIFY messages on exten/device state changes.

Postpone destruction of old subscriptions until all already queued tasks
that may be using old subscriptions have completed.

ASTERISK-29906

Change-Id: I96582aad3a26515ca73a8460ee6756f56f6ba23b

* Added processing for the 'confidence' element.
* Added documentation to some APIs.
* removed a lot of complex code related to the very-off-nominal
  case of needing to process multiple location info sources.
* Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes
  one eprofile instead of a datastore of multiples.
* Plugged a huge leak in XML processing that arose from
  insufficient documentation by the libxml/libxslt authors.
* Refactored stylesheets to be more efficient.
* Renamed 'profile_action' to 'profile_precedence' to better
  reflect it's purpose.
* Added the config option for 'allow_routing_use' which
  sets the value of the 'Geolocation-Routing' header.
* Removed the GeolocProfileCreate and GeolocProfileDelete
  dialplan apps.
* Changed the GEOLOC_PROFILE dialplan function as follows:
  * Removed the 'profile' argument.
  * Automatically create a profile if it doesn't exist.
  * Delete a profile if 'inheritable' is set to no.
* Fixed various bugs and leaks
* Updated Asterisk WiKi documentation.

ASTERISK-30167

Change-Id: If38c23f26228e96165be161c2f5e849cb8e16fa0

Change-Id: I68ba22db0a69d9e2eabcc2141b48a2395f7f1a23

Adjusts some logging levels to be more or less important,
that is more prominent when actual problems occur and less
prominent for less noteworthy things.

ASTERISK-30153 #close

Change-Id: Ifc8f7df427aa018627db462125ae744986d3261b