AST_CONTROL_FLASH isn't accounted for in a switch statement in file.c
where it should be ignored. Adding this to the switch ensures a
warning isn't thrown on RFC2833 flash events, since nothing's amiss.

ASTERISK-29372

Change-Id: I4fa549bfb7ba1894a4044de999ea124877422fbc

ASTERISK-29358 #close

Change-Id: I050daff67066873df4e8fc7f4bd977c1ca06e647

STIR/SHAKEN encodes using base64 URL format. Currently, we just use
base64. New functions have been added that convert to and from base64
encoding.

The origid field should also be an UUID. This means there's no reason to
have it as an option in stir_shaken.conf, as we can simply generate one
when creating the Identity header.

https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021

Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c

We're getting the serial number of the certificate from openssl and
freeing it with ast_free(), but it needs to be freed with OPENSSL_free()
instead. Now we duplicate the string and free the one from openssl with
OPENSSL_free(), which means we can still use ast_free() on the returned
string.

https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021

Change-Id: Ia6e1a4028c1933a0e1d204b769ebb9f5a11f00ab

During OpenSIPit, we found out that the public certificates must be of
type X.509. When reading in public keys, we use the corresponding X.509
functions now.

We also discovered that we needed a better naming scheme for the
certificates since certificates with the same name would cause issues
(overwriting certs, etc.). Now when we download a public certificate, we
get the serial number from it and use that as the name of the cached
certificate.

The configuration option public_key_url in stir_shaken.conf has also
been renamed to public_cert_url, which better describes what the option
is for.

https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021

Change-Id: Ia00b20835f5f976e3603797f2f2fb19672d8114d

Enhancements:

 * The MessageSend dialplan application now takes an optional
   third argument that can set the message's "To" field on
   outgoing messages.  It's an alternative to using the
   MESSAGE(to) dialplan function.

   NOTE: No channel driver currently implements this field.  A
   follow-on commit for res_pjsip_messaging will implement it for
   the chan_pjsip channel driver.

 * To prevent confusion with the first argument, currently named
   "to", it's been renamed to "destination". Its function,
   creating the request URI, hasn't changed.

 * The documentation for MessageSend was updated to be
   more clear about the parameters and how they interact
   the MESSAGE() dialplan function.

 * With the rename of MessageSend's first parameter, and the fact
   that message.c references <info> elements in chan_sip.c,
   res_pjsip_messaging.c and res_xmpp, they each needed
   documentation updates to use MessageDestinationInfo instead of
   MessageToInfo.

 * appdocsxml.dtd was updated to include a missing element
   declaration for "dataType".  This was showing up as an error
   in Eclipse's dtd editor.

 * Despite the changes in this commit, there should be
   no impact to current users of MessageSend.

Change-Id: I6fb5b569657a02866a66ea352fd53d30d8ac965a

Change-Id: Idbd61ff77545f4a78b06a5064b55112e774b70e6

When a stream topology is provided to chan_local when dialing
it filters the audio formats down. This operation did not skip
streams which were removed (that have no formats) resulting in
calling being aborted.

This change causes such streams to be skipped.

ASTERISK-29407

Change-Id: I1de8b98727cb2d10f4bc287da0b5fdcb381addd6

We assume that a timestamp value of 0 represents an 'uninitialized'
timestamp, but 0 is a valid value. Add a simple wrapper to be able to
differentiate between whether the value is set or not.

This also removes the fix for ASTERISK~28812 which should not be
needed if we are checking the last timestamp appropriately.

ASTERISK-29030 #close

Change-Id: Ie70d657d580d9a1f2877e25a6ef161c5ad761cf7

ASTERISK-27477 #close

Change-Id: I68f6715bba92a525149e35d142a49377a34a1193

Up/down sampling changes the number of samples produced by a translation.
This must be taken into account when checking the codec's buffer size.

ASTERISK-29328

Change-Id: I9aebe2f8788e00321a7f5c47aa97c617f39e9055

Some versions of SVN seemingly don't follow the redirect
to https.

Change-Id: Ia7c76c18cb620bcf56f08e1211a7d80d321fe253

Change-Id: I2f76867ce02ec611964925159be099de83346e38

There is a possibility, when bridge_channel_write_frame() is
called, that the bridge_channel->chan will be NULL.  The first
thing bridge_channel_write_frame() does though is call
ast_channel_is_multistream() which had no check for a NULL
channel and therefore caused a segfault. Since it's still
possible for bridge_channel_write_frame() to write the frame to
the other channels in the bridge, we don't want to bail before we
call ast_channel_is_multistream() but we can just skip the
multi-channel stuff.  So...

bridge_channel_write_frame() only calls ast_channel_is_multistream()
if bridge_channel->chan is not NULL.

As a safety measure, ast_channel_is_multistream() now returns
false if the supplied channel is NULL.

ASTERISK-29379
Reported-by: Vyrva Igor
Reported-by: Ross Beer

Change-Id: Idfe62dbea8c69813ecfd58e113a6620dc42352ce

Only use an XPath query once per module, then just navigate the DOM for
everything else.

Change-Id: Ia0336a7185f9180ccba4b6f631a00f9a22a36e92

The channels, bridges and endpoints scrape functions were
grabbing their respective global containers, getting the
count of entries, allocating metric arrays based on
that count, then iterating over the container.  If the
global container had new objects added after the count
was taken and the metric arrays were allocated, we'd run
out of metric entries and attempt to write past the end
of the arrays.

Now each of the scape functions clone their respective
global containers and all operations are done on the
clone.  Since the clone is stable between getting the
count and iterating over it, we can't run past the end
of the metrics array.

ASTERISK-29130
Reported-By: Francisco Correia
Reported-By: BJ Weschke
Reported-By: Sébastien Duthil

Change-Id: If0c8e40853bc0e9429f2ba9c7f5f358d90c311af

Using the information from the MODULEINFO XML we can
now output useful information at the end of module
loading for deprecated modules. This includes the
version it was deprecated in, the version it will be
removed in, and the replacement if available.

ASTERISK-29339

Change-Id: I2080dab97d2186be94c421b41dabf6d79a11611a

For some input to the standard deviation algorithm extremely large,
and wrong numbers were being calculated.

This patch uses a new formula for correctly calculating both the
running mean and standard deviation for the given inputs.

ASTERISK-29364 #close

Change-Id: Ibc6e18be41c28bed3fde06d612607acc3fbd621f

The calculated minimum lost packets represents the lowest number of
lost packets missed during an RTCP report interval. Zero of course
is the lowest, but the idea is that this value contain the lowest
number of lost packets once some have been missed.

This patch checks to make sure the number of lost packets over an
interval is not zero before checking and setting the minimum value.

Also, this patch updates the rtp lost packet test to check for
packet loss over several reports vs one.

Change-Id: I07d6e21cec61e289c2326138d6bcbcb3c3d5e008

This patch makes the drop_packets_data object static.

Change-Id: If4f9b21fa0c47d41a35b6b05941d978efb4da87b

Flash in RTP is conveyed the same as DTMF, just with a
specific digit. In Asterisk however we do flash as a
single control frame.

This change makes it so that only on end do we provide
the flash control frame to the core. Previously we would
provide a flash control frame on both begin and end,
causing flash to work improperly.

ASTERISK-29373

Change-Id: I1accd9c6e859811336e670e698bd8bd124f33226

This patch makes it so when Asterisk is compiled in DEVMODE a CLI
command is available that allows someone to drop incoming RTP
packets. The command allows for dropping of packets once, or on a
timed interval (e.g. drop 10 packets every 5 seconds). A user can
also specify to drop packets by IP address.

Change-Id: I25fa7ae9bad6ed68e273bbcccf0ee51cae6e7024

Change-Id: I058af496021ff870ccec2d8cbade637b348ab80b

Added a TIME_UNIT enumeration, and a function that converts a
string to one of the enumerated values. Also, added functions
that create and initialize a timeval object using a specified
value, and unit type.

Change-Id: Ic31a1c3262a44f77a5ef78bfc85dcf69a8d47392

ASTERISK-28356 #close

Change-Id: I53a1bfdd3113d620bea88349019173a2f3f0ae39

I missed the changes in 18 and master in the previous review.

ASTERISK-24434 #close

Change-Id: Ieb132b2a998ce96daa9c9acf26535a974b895876

Change-Id: Iff0e713f2120d8dce8e1e26924b99ed17f9d9dff

ASTERISK-24631 #close

Change-Id: I8bf8776906a72ee02f24de6a85345940b9ff6b6f

Also removed the sample documentation, and some oddly-placed
documentation about the timeout argument to the Queue() application
itself. There is a large section on the timeout behavior below.

ASTERISK-26614 #close

Change-Id: I8f84e8304b50305b7c4cba2d9787a5d77c3a6217

The 'core' console (ie: asterisk -c) does read logger.conf and does
use the dateformat= option.

Whereas 'remote' consoles (ie: asterisk -r -T) does not read logger.conf
and uses a hard coded dateformat option for printing received verbose messages:
  main/logger.c: static char dateformat[256] = "%b %e %T"

This change will load logger.conf for each remote console session and
use the dateformat= option to set the per-line timestamp for verbose messages

Change-Id: I3ea10990dbd920e9f7ce8ff771bc65aa7f4ea8c1
ASTERISK-25358: #close
Reported-by: Igor Liferenko

ASTERISK-27542 #close

Change-Id: If0b9719380a25533d2aed1053cff845dc3a4854a

When the check for equal topologies was added to reschedule_reinvite()
it was assumed that both the pending and active media states would
actually have non-NULL topologies.  We since discovered this isn't
the case.

We now only test for equal topologies if both media states have
non-NULL topologies.  The logic had to be rearranged a bit to make
sure that we cloned the media states if their topologies were
non-NULL but weren't equal.

ASTERISK-29215

Change-Id: I61313cca7fc571144338aac826091791b87b6e17

If a queue member was updated with the same status multiple
times each time a QueueMemberStatus event would be sent
which would be a duplicate of the previous.

This change makes it so that the QueueMemberStatus event is
only sent if the status actually changes.

ASTERISK-29355

Change-Id: I580c60d992a0a8f2bea8b91c868771b3b490d116

When using the ast_unreal_lock_all function no channel
locks can be held before calling it.

This change unlocks the channel that indicate was
called on before doing so and then relocks it afterwards.

ASTERISK-29035

Change-Id: Id65016201b5f9c9519a216e250f9101c629e19e9

Some configuration items for a transport do not result in
the underlying transport changing, but instead are just
state we keep ourselves and use. It is perfectly reasonable
to change these items.

These include local_net and external_* information.

ASTERISK-29354

Change-Id: I027857ccfe4419f460243e562b5f098434b3d43a

ASTERISK-29348

Change-Id: I77e3466435f5a51a57538b29addb68d811af238d
Signed-off-by: Jaco Kroon <jaco@uls.co.za>

When an SSRC change occurs the timestamps are likely
to change as well. As a result we need to reset the
timestamp mapping done in the calc_rxstamp function
so that they map properly from timestamp to real
time.

This previously occurred but due to packet
retransmission support the explicit setting
of the marker bit was not effective.

ASTERISK-29352

Change-Id: I2d4c8f93ea24abc1030196706de2d70facf05a5a

The "deprecated_in" and "removed_in" information can now be
set in MODULEINFO for a module and is then displayed in
menuselect so users can be aware of when a module is slated
to be deprecated and then removed.

ASTERISK-29337

Change-Id: I6952889cf08e0e9e99cf8b43f99b3cef4688087a

Some modules have a different support level documented in their
MODULEINFO XML and Asterisk module definition. This change
brings the two in sync for the modules which were not matching.

ASTERISK-29336

Change-Id: If2f819103d4a271e2e0624ef4db365e897fa3d35