When a connected line update is sent to an endpoint we do not request
a specific stream topology to be used. Previously this resulted in the
configured stream topology being used which may actually differ from the
currently negotiated topology. PJSIP is helpful in this regard in that
it will fill in any missing streams with removed ones. This results in
our own state not matching the SDP, though, and we do not apply the
negotiated SDP.

This change tweaks the code to use the actively negotiated stream
topology if it is present with a fallback to the configured one. This
results in the SDP and the state having matching information and the
world is happy.

ASTERISK*27397

Change-Id: I7a57117f0183479e6884b7bf3a53bb8c7464f604

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

Change-Id: I227e2d10c0035bbfa1c6e46ae2318fd1122d8420

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: 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

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

* Add the channel name to diagnostic messages so you will know which
channel failed to transfer.

* Promoted some debug messages to verbose 4 messages.

ASTERISK-27449 #close

Change-Id: Idac66b7628c99379cc9269158377fd87dc97a880

The SuccessfulAuth using_password field was declared as a pointer to a
uint32_t when the field was later read as a uint32_t value.  This resulted
in unnecessary casts and a non-portable field value reinterpret in
main/security_events.c:add_json_object().  i.e., It would work on a 32 bit
architecture but not on a 64 bit big endian architecture.

Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935

More complicated direct media reinvite negotiations can result in longer
delays before direct media flows.  The strictrtp learning timeout time
was too short.  One log showed that the first RTP packet came in just
after three seconds.

* Increase the strictrtp learning timeout time from 1.5 to 5 seconds.

ASTERISK-27453

Change-Id: Ic5e711164cbb91b4d1c1e40c83697755640f138c

With Asterisk 12 (commit 866d9681), the default of "icesupport" changed to
- "yes" in the module "res_rtp_asterisk" and
- "no" in the module "chan_sip".
The latter was reflected in the sample configuration file for "sip.conf". The
former did not make it into "rtp.conf.sample".

ASTERISK-20643

Change-Id: I2a2e0a900455d0767a99ea576e30adc6d7608a36