When the RTCP code was transitioned over to Stasis a code change
was made to keep track of how many reports are present. This count
controlled where report blocks were placed in the RTCP report.

If a compound RTCP packet was received this logic would incorrectly
place a report block in the wrong location resulting in a write
to an invalid location.

This change removes this counting logic and always places the report
block at the first position. If in the future multiple reports are
supported the logic can be extended but for now keeping a count
serves no purpose.

ASTERISK-27382
ASTERISK-27429

Change-Id: Iad6c8a9985c4b608ef493e19c421211615485116

This is a partial fix for ASTERISK~25817 but does not address the
comments regarding RFC 5626.

Change-Id: I227e2d10c0035bbfa1c6e46ae2318fd1122d8420

Stripping the DNID in a SIP dial string can result in attempting to call
the argument parsing macros on an empty string, causing a crash.

ASTERISK-26131 #close
Reported by: Dwayne Hubbard
Patches:
	dw-asterisk-master-dnid-crash.patch (license #6257) patch
	uploaded by Dwayne Hubbard

Change-Id: Ib84c1f740a9ec0539d582b09d847fc85ddca1c5e

This patch does three things associated with the initial incoming INVITE
request URI.

1) Add access to the full initial incoming INVITE request URI.

2) We were not setting DNID on incoming PJSIP channels.  The DNID is the
user portion of the initial incoming INVITE Request-URI.  The value is
accessed by reading CALLERID(dnid).

3) Fix CHANNEL(pjsip,target_uri) documentation.

* The initial incoming INVITE request URI is now available using
CHANNEL(pjsip,request_uri).

* Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the
initial incoming INVITE request URI user portion.

* CHANNEL(pjsip,target_uri) now correctly documents that the target URI is
the contact URI.

* Refactored print_escaped_uri() out of channel_read_pjsip() to handle
pjsip_uri_print() error condition when the buffer is too small.

ASTERISK-27478

Change-Id: I512e60d1f162395c946451becb37af3333337b33

Support for these protocols was added in the same commit as the 'proto'
field, so we can safely use the same ./configure check.

For reference: https://trac.pjsip.org/repos/changeset/4968

Change-Id: Icf4975d785d6bfb8f30ac7ffa695a0adf9382dac

Change-Id: I51f6945c4023cb93fc7b87be5ab4c50e9e6ee27d

Handle CLI initialization before any processing occurs.

Change-Id: I598b911d2e409214bbdfd0ba0882be1d602d221c

Change-Id: Ib8d45bbdfbda81e65045f6dff874d189b74e5471

ASTERISK-27475 #close

Change-Id: If7384bc6ed002ef140dec69798d14c52b7cfd800

Use the new ast_cli_completion_add() function to improve completion
performance for commands like 'pjsip show endpoint.'

Change-Id: I76d802294d2ac1766110dc75f7d117c8541ce348

Using the LIKE operator requires a full table scan of 'astdb', whereas a
comparison operation is able to use the primary key index.

This patch adds a new function to the AstDB API for quick prefix matches
and updates res_sorcery_astdb to utilize it. This showed substantial
performance improvement in my test environment.

Related to ASTERISK~26806, but does not completely resolve it.

Change-Id: I7d37f9ba2aea139dabf2ca72d31fbe34bd9b2fa1

Optimize resource_name_match.  This change eliminates use of
ast_strdupa, instead verifying that both basename's are the same length,
then using strncasecmp.

Change-Id: I477275c0e954c99d74be5abfc8bb6545b04e5a3d

Change-Id: I25348c386a222bb704aff07f54375108a6402906

There are many places in the code base where we ignore the return value
of fcntl() when getting/setting file descriptior flags. This patch
introduces a convenience function that allows setting or clearing file
descriptor flags and will also log an error on failure for later
analysis.

Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7

res_stasis was missing AST_MODFLAG_LOAD_ORDER.  Set res_stasis and
res_speech to start at (AST_MODPRI_APP_DEPEND - 1) so they are ready for
dependent modules.

Change-Id: I27f4f3810a95b6be8a5bfbf62be2ace6bfab6ff3

For both dynamic and static contacts it was possible that potential AOR
changes were not being applied to all contacts. This was because the qualify
and schedule code was only retrieving AOR's, and contacts with frequencies
greater than zero.

For instance the following could happen: and AOR/contact has a frequency of 5,
it then gets set to 0, and then a reload occurs. All scheduled OPTIONS are
stopped, a list of AOR's is retrieved with frequency > 0, but none are
selected since in this scenario all are 0. The contact for the one previously
set to 5 though does not get updated, so it's status remains "AVAILABLE".

This patch makes it so all contacts (static and dynamic) are selected, and
appropriately updated if need be.

ASTERISK-27467 #close

Change-Id: I7a920170f89c683af9505d4723a44fc6841decdb

Dynamic contacts were not being properly updated on reload. As a matter of
fact any changes to the AOR that a dynamic contact was associated with were
not being applied.

On reload, this patch makes it so for each dynamic contact, the associated
AOR is now retrieved and the AOR's fields are applied to the contact.

ASTERISK-27467

Change-Id: I8e3165dc6a745218c1c9db837f77fafa0516985d

Change-Id: I1e5eef4029cba56e33d786c5a5ade8091e531a1e

The sounds index is rebuilt each time a format is registered or
unregistered.  This causes the index to be repeatedly rebuilt during
startup and shutdown.

This patch significantly reduces the work done by delaying sound index
initialization until after modules are loaded.  This way a reindex only
occurs if a format module is loaded after startup.  We also skip
reindexing when format modules are unloaded during shutdown.

Change-Id: I585fd6ee04200612ab1490dc804f76805f89cf0a

Setting channel variables with the AMI Originate action caused a deadlock
when you set CDR(amaflags) or CDR(accountcode).  This path has the channel
locked when the CDR function is called.  The CDR function then
synchronously passes the job to a stasis thread.  The stasis handling
function then attempts to lock the channel.  Deadlock results.

* Avoid deadlock by making the CDR function handle setting amaflags and
accountcode directly on the channel rather than passing it off to the CDR
processing code under a stasis thread to do it.

* Made the CHANNEL function and the CDR function process amaflags the same
way.

* Fixed referencing the wrong message type in cdr_prop_write().

ASTERISK-27460

Change-Id: I5eacb47586bc0b8f8ff76a19bd92d1dc38b75e8f

This eliminates some wasteful operations in media_index startup.

* Replace statically set string-fields with char[0].
* Eliminate pointless RAII_VAR's.
* alloc_variant: Avoid pointless ao2_find on new info->variant.
* Stop trying find_variant before alloc_variant.
* process_media_file: replace ast_str with ast_asprintf.  This avoids
  reallocation of file_id_str.

Overall sounds_index.c is about 27% of Asterisk startup time when using
sample configs.  This patch reduces it to 20%.  This is a half-fix.  The
real problem is that the media_index is regenerated repeatedly - 68
times in my test.

Change-Id: Ia50b752f8efb356f852b05c4be495a6631af8652

* Added start DTMF transfer verbose messages.
* Made associated transfer messages use a similar message format.
* Adjusted message verbose level as requested by initial reporter.

ASTERISK-27449

Change-Id: I2045714586414b3c5ef1f3cc56c1c4af4b31f551