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

In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the
fallback use of the transport's bind address solve problems sending
media on systems that cannot send ipv4 packets on ipv6 sockets, and
certain other situations. This change extends both of these behaviors
to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific
problems on these systems, introducing a new option
endpoint/t38_bind_udptl_to_media_address.

ASTERISK-29402

Change-Id: I87220c0e9cdd2fe9d156846cb906debe08c63557

If the terminator character is not explicitly specified
and an indications tone is used for reading a digit,
there is no null pointer check so Asterisk crashes.
This prevents null usage from occuring.

ASTERISK-29673 #close

Change-Id: Ie941833e123c3dbfb88371b5de5edbbe065514ac

Add missing reference decrement in rtp_deallocate_transport()

ASTERISK-29671

Change-Id: I8d22dbedb90e8dade0829b7a28372f404b07caa9

The current versions do not support future dates in all say application when using the 'Q' or 'q' format parameter and says "today" for everything that is greater than today

ASTERISK-29637

Change-Id: I1fb1cef0ce3c18d87b1fc94ea309d13bc344af02

The behavior of max_contacts and remove_existing are connected.  If
remove_existing is enabled, the soonest expiring contacts are removed.
This may occur when there is an unavailable contact.  Similarly,
when remove_existing is not enabled, registrations from good
endpoints are rejected in favor of retaining unavailable contacts.

This commit adds a new AOR option remove_unavailable, and the effect
of this setting will depend on remove_existing.  If remove_existing
is set to no, we will still remove unavailable contacts when they
exceed max_contacts, if there are any. If remove_existing is set to
yes, we will prioritize the removal of unavailable contacts before
those that are expiring soonest.

ASTERISK-29525

Change-Id: Ia2711b08f2b4d1177411b1be23e970d7fdff5784

When listing bridges we go through the ones present in
ARI, get their snapshot, turn it into JSON, and add it
to the payload we ultimately return.

An invisible "dial bridge" exists within ARI that would
also try to be added to this payload if the channel
"create" and "dial" routes were used. This would ultimately
fail due to invisible bridges having no snapshot
resulting in the listing of bridges failing.

This change makes it so that the listing of bridges
ignores invisible ones.

ASTERISK-29668

Change-Id: I14fa4b589b4657d1c2a5226b0f527f45a0cd370a

Allows multiple mailboxes to be specified for VMCOUNT
instead of just one.

ASTERISK-29661 #close

Change-Id: I9108528300795fd5b607efa9d4dd7b74be031813

The MessageSend AMI action has been updated to allow the Destination
and the To addresses to be provided separately. This brings the
MessageSend manager command in line with the capabilities of the
MessageSend dialplan application.

ASTERISK-29663 #close

Change-Id: I8513168d3e189a9fed88aaab6f5547ccb50d332c

Adds a function to check for the existence of a channel by
name or by UNIQUEID.

ASTERISK-29656 #close

Change-Id: Ib464e9eb6e13dc683a846286798fecff4fd943cb

Previously, if custom hints were used with the hint:
format in app_queue, when device state changes occured,
app_queue would only do a literal string comparison of
the context used for the hint in app_queue and the context
of the hint which just changed state. This caused hints
to not update and become stale if the context associated
with the agent included the context which actually changes
state, essentially completely breaking device state for
any such agents defined in this manner.

This fix adds an additional check to ensure that included
contexts are also compared against the context which changed
state, so that the behavior is correct no matter whether the
context is specified to app_queue directly or indirectly.

ASTERISK-29578 #close

Change-Id: I8caf2f8da8157ef3d9ea71a8568c1eec95592b78

Rather than stripping parameters from Content-Type headers before
comparison, first try to compare the whole string. If no match is
found, strip the parameters and try that way.

ASTERISK-29275 #close

Change-Id: I2963c8ecbb3a9605b78b6421c415108d77a66a0f

Adds the ability for users to log to custom log levels
by providing custom log level names in logger.conf. Also
adds a logger show levels CLI command.

ASTERISK-29529

Change-Id: If082703cf81a436ae5a565c75225fa8c0554b702

No functional changes.

Change-Id: I46514152c0af67f395526374aaa847ccd6a85378

Some code has been added referencing symbols defined in a block
protected by #ifdef HAVE_PJPROJECT. Protect those code parts in
ifdef blocks too.

ASTERISK-29660

Change-Id: Ib18d4392d51ac80ca5481dabf6e498a4e3e49e6f

An issue was found where a particular manufacturer's phones add a
trailing space to the end of the rtpmap attribute when specifying
a payload type that has a "param" after the format name and clock
rate. For example:

a=rtpmap:120 opus/48000/2 \r\n

Because pjmedia_sdp_attr_get_rtpmap currently takes everything after
the second '/' up to the line end as the param, the space is
included in future comparisons, which then fail if the param being
compared to doesn't also have the space.

We now use pj_scan_get() to parse the param part of rtpmap so
trailing whitespace is automatically stripped.

ASTERISK-29654

Change-Id: Ibd0a4e243a69cde7ba9312275b13ab62ab86bc1b

In new mpg123 versions (since 1.26) the default output is 32 bits
Asterisk expects the output in 16 bits, so we force the output to be on 16 bits.
It will work wit new and old versions of mpg123.
Thanks Thomas Orgis <thomas-forum@orgis.org> for giving the key!

ASTERISK-29635 #close

Change-Id: I88c7740118b5af4e895bd8b765b68ed5c11fc816

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 a SendMF application and PlayMF manager
event to send arbitrary R1 MF tones on the
current or specified channel.

ASTERISK-29496

Change-Id: I5d89afdbccee3f86cc702ed96d882f3d351327a4

Previously, the error emitted when app_stack tries
to branch to a dialplan location that doesn't exist
has included only the information about the attempted
branch in the error log. This adds the current location
as well so users can see where the branch failed in
the logs.

ASTERISK-29626

Change-Id: Ia23502ab2ad21485a1ac74295063a8f25a6df5ce

Change-Id: I9a3a978b2f818be464e062d97b93831b127ef28c

Fixed the external media creation handle to handle the 'data' option correctly.

ASTERISK-29629

Change-Id: I22e57fe8ebf3d3e08fb2121aa4a8a52cc62e8129

Adds the STRBETWEEN function, which can be used to insert a
substring between each character in a string. For instance,
this can be used to insert pauses between DTMF tones in a
string of digits.

ASTERISK-29627

Change-Id: Ice23009d4a8e9bb9718d2b2301d405567087d258

We can't rely on RAII_VAR(...) to properly clean up data that is
allocated within a loop.

ASTERISK-27176 #close

Change-Id: Ib575616101230c4f603519114ec62ebf3936882c

Adds the DIRNAME and BASENAME functions, which are
wrappers around the corresponding C library functions.
These can be used to safely and conveniently work with
file paths and names in the dialplan.

ASTERISK-29628 #close

Change-Id: Id3aeb907f65c0ff96b6e57751ff0cb49d61db7f3

Up until now, all of the logic used to translate
arguments to the Say applications has been
directly coupled to playback, preventing other
modules from using this logic.

This refactors code in say.c and adds a SAYFILES
function that can be used to retrieve the file
names that would be played. These can then be
used in other applications or for other purposes.

Additionally, a SayMoney application and a SayOrdinal
application are added. Both SayOrdinal and SayNumber
are also expanded to support integers greater than
one billion.

ASTERISK-29531

Change-Id: If9718c89353b8e153d84add3cc4637b79585db19

dsp.c contains arbitrary tone detection functionality
which is currently only used for fax tone recognition.
This change makes this functionality publicly
accessible so that other modules can take advantage
of this.

Additionally, a WaitForTone and TONE_DETECT app and
function are included to allow users to do their
own tone detection operations in the dialplan.

ASTERISK-29546

Change-Id: Ie38c395000f4fd4d04e942e8658e177f8f499b26

With gcc 11, res/res_snmp.c and res/snmp/agent.c need the
-fPIC option added to its _ASTCFLAGS.

ASTERISK-29634

Change-Id: I34649c85e075fd954e578378fabf798c3f038f50

There is an option to silence voicemail instructions but it does not
take into consideration if a recorded greeting exists or not. Add a
new 'S' option that does that.

ASTERISK-29632 #close

Change-Id: I03f2f043a9beb9d99deab302247e2a8686066fb4

ncurses 6.1 introduced an extended number format for terminfo files
which the terminfo parsing in Asterisk is not able to parse. This
results in some TERM values that do support color (screen-256color on
Ubuntu 20.04 for example) to not get a color console.

ASTERISK-29630 #close

Change-Id: I27a4fcfab502219924af2d6b1c46feba92903cb3

When compiled without extended srtp crypto suites also disable parsing
these from received SDP. This prevents using these, as some client
implementations are not stable.

ASTERISK-29625

Change-Id: I7dafb29be1cdaabdc984002573f4bea87520533a

IPv6 nameserver addresses are stored in different part of the
__res_state structure, so look there if we appear to have support for
it.

ASTERISK-28004 #close

Change-Id: I67067077d8a406ee996664518d9c8fbf11f6977d

There are conditions under which a failure to change topology
is expected so there's no need to print an ERROR message.

ASTERISK-29618
Reported by: Alexander

Change-Id: Idc168b8588e018bf3a23769f08c4ad646086d481

Fixed ARI external media handler to accept body parameters.

ASTERISK-29622

Change-Id: I49509c48a6cbc0fb4165bfa4f834b5e8b9ace20d

There are 3 separate changes here but they are all closely related:

* Only try to set matchfield attributes on 'field' nodes

* We need to adjust how we treat the category pointer based on the
  value of the category_match, to avoid memory corruption. We now
  generate a regex-like string when match types other than
  ACO_WHITELIST and ACO_BLACKLIST are used.

* Switch app_agent_pool from ACO_BLACKLIST_ARRAY to
  ACO_BLACKLIST_EXACT since we only have one category we need to
  ignore, not two.

ASTERISK-29614 #close

Change-Id: I7be7bdb1bb9814f942bc6bb4fdd0a55a7b7efe1e

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

ASTERISK-29605 #close

Change-Id: Iaeacdf6ccde18eaff7f776a0f49fee87dcb549d2

Currently pbx_ael does not check if a reload is currently pending
before proceeding with a reload. This can cause multiple threads to
operate at the same time on what should be mutex protected data. This
change adds protection to reloading to ensure only one ael reload is
executing at a time.

ASTERISK-29609 #close

Change-Id: I5ed392ad226f6e4e7696ad742076d3e45c57af35

Allows for the digit # to be read as a digit,
just like any other DTMF digit, as opposed to
forcing it to be used as an end of input
indicator. The default behavior remains
unchanged.

ASTERISK-18454 #close

Change-Id: I3033432adb9d296ad227e76b540b8b4a2417665b

This allows the STUN server to change its IP address without having to
reload the res_rtp_asterisk module.

The refresh of the name resolution occurs first when the module is
loaded, then recurringly, slightly after the previous DNS answer TTL
expires.

ASTERISK-29508 #close

Change-Id: I7955a046293f913ba121bbd82153b04439e3465f

The attended transfer feature will emit a warning if the user
cancels the transfer or the attended transfer doesn't complete
for any reason. Changes the warning to a verbose message,
since nothing is actually wrong here.

ASTERISK-29612 #close

Change-Id: I64c93cdb21360a0a8d45e9cb6db3af8168f66e6d