This allows asterisk to be compiled with MALLOC_DEBUG to load modules
built without MALLOC_DEBUG.  Now pre-compiled third-party modules will
still work regardless of MALLOC_DEBUG being enabled or not.

Change-Id: Ic07ad80b2c2df894db984cf27b16a69383ce0e10

The pool cache gets in the way of finding use after free errors of memory
pool contents.  Tools like valgrind and MALLOC_DEBUG don't know when a
pool is released because it gets put into the cache instead of being
freed.

* Added the "cache_pools" option to pjproject.conf.  Disabling the option
helps track down pool content mismanagement when using valgrind or
MALLOC_DEBUG.  The cache gets in the way of determining if the pool
contents are used after free and who freed it.

To disable the pool caching simply disable the cache_pools option in
pjproject.conf and restart Asterisk.

Sample pjproject.conf setting:
[startup]
cache_pools=no

* Made current users of the caching pool factory initialization and
destruction calls call common routines to create and destroy cached pools.

ASTERISK-27704

Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828

Change-Id: Id52f719078a65c4b2eee7ab99d761eba6b6aed94

Since res_pjsip_transport_management provides several attack
mitigation features, its functionality moved to res_pjsip and
this module has been removed.  This way the features will always
be available if res_pjsip is loaded.

ASTERISK-27618
Reported By: Sandro Gauci

Change-Id: I21a2d33d9dda001452ea040d350d7a075f9acf0d

* app_fax (replaced by res_fax).
* res_config_sqlite (replaced by res_config_sqlite3).
* res_monitor (replaced by app_mixmonitor).

This is related to ASTERISK~23657 but does not resolve that ticket.
Resolving that ticket would require complete removal of res_monitor.

ASTERISK-27671 #close

Change-Id: I16a3edd61fc1abd4a7b2e9357693ed663f62dd49

This removes the embedded copy of editline from the Asterisk source
tree, making a system copy of libedit mandatory in Asterisk 16+.

ASTERISK-27634 #close

Change-Id: Iedb64ad92acb78419f3caefedaa2bb7cd2a1a33f

* Made the AMI ConfbridgeList action's ConfbridgeList events output all
the standard channel snapshot headers instead of a few hand-coded channel
snapshot headers.  The benefit is that the CallerIDName gets disruptive
characters like CR, LF, Tab, and a few others escaped.  However, an empty
CallerIDName is now output as "<unknown>" instead of "<no name>".

ASTERISK-27651

Change-Id: Iaf7d54a9d40194c2db060bc9b4979fab6720d977

ASTERISK-27651

Change-Id: Idef2ca54d242d1b894efd3fc7b360bc6fd5bdc34

Verified nothing in the testsuite lists this module as a dependency.

Change-Id: I90c7d52c7e15e85fde3389d5eaccb05b97848813

In an earlier release, inbound registrations on a reliable transport
were pruned on Asterisk restart since the TCP connection would have
been torn down and become unusable when Asterisk stopped.  This same
process is now also applied to inbound subscriptions.

Also fixed issues in res_pjsip_registrar where it wasn't handling the
monitoring correctly when multiple registrations came in over the same
transport.

To accomplish this, the pjsip_transport_event feature needed to
be refactored to allow multiple monitors (multiple subcriptions or
registrations from the same endpoint) to exist on the same transport.
Since this changed the API, any external modules that may have used the
transport monitor feature (highly unlikey) will need to be changed.

ASTERISK-27612
Reported by: Ross Beer

Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36

ASTERISK-27581

Change-Id: If6af275764741a11030f0a4fd324fa29b376d74e

Each time the dial plan is reloaded, a lot of logs like these are generated:
"Added extension 'XXXXX' priority 1 to YYYYYYYYYYY"
This patch changes the log level for those logs.

ASTERISK-27084

Change-Id: I5662902161c50890997ddc56835d4cafb456c529

This patch adds the ability to configure a prompt which will be read
to the "winner" who pressed 1 (or the configured value) and received
the call.

ASTERISK-24372 #close

Change-Id: I6ec1c6c883347f7d1e1f597189544993c8d65272

The type=identify endpoint identification method can match by IP address
and by SIP header.  However, the SIP header matching has limited
usefulness because you cannot specify the SIP header matching priority
relative to the IP address matching.  All the matching happens at the same
priority and the order of evaluating the identify sections is
indeterminate.  e.g., If you had two type=identify sections where one
matches by IP address for endpoint alice and the other matches by SIP
header for endpoint bob then you couldn't predict which endpoint is
matched when a request comes in that matches both.

* Extract the SIP header matching criteria into its own "header" endpoint
identification method so the user can specify the relative priority of the
SIP header and the IP address matching criteria in the global
endpoint_identifier_order option.  The "ip" endpoint identification method
now only matches by IP address.

ASTERISK-27491

Change-Id: I9df142a575b7e1e3471b7cda5d3ea156cef08095

There has been an open issue against cdr_syslog (ASTERISK~14441) about
a race condition for 7.5 years that has never been addressed. Because
this module is effectively unmaintained and currently broken, there is
no sense in keeping it around.

If logging CDRs to syslog is a desirable feature, it would probably be
better to write the logs directly to the syslog server via socket
instead of using the facilities provided by openlog/syslog/closelog.
Doing so would address the race condition referenced in the associated
issue.

Change-Id: Ic77b94cd97f355a9cf5b1d3f3444964a6e0ba5dc

Add an AMI action which provides information on all
configured Auths.

ASTERISK-27547

Change-Id: I1a88a75b38a2b1dd9d1de6c0307b20a3f584c817

Add an AMI action which provides information on all
configured AORs.

ASTERISK-27537

Change-Id: If8b990a00909e5b6c0f04a3b8dccd9903dc445eb

This patch adds the ability to set the wrapuptime on the queue member
config.

When the option is set the wrapuptime on the queue member is used instead
of the queue's wrapuptime.

ASTERISK-27483 #close

Change-Id: I11c85809537f974eb44dc5bbf82bcedd8a458902

Log a message to security events when an INVITE is received to an
invalid extension.

ASTERISK-25869 #close

Change-Id: I0da40cd7c2206c825c2f0d4e172275df331fcc8f

A couple of places were setting the status to "UNKNOWN" when qualifies were
being disabled. Instead this should be set to the "CREATED" status that
represents when a contact is given (uri available), but the qualify frequency
is set to zero so we don't know the status.

This patch updates the relevant places with "CREATED". It also updates the
"CREATED" status description (value shown in CLI/AMI/ARI output) to a value
of "NonQualified"/"NonQual" as this description is hopefully less confusing.

ASTERISK-27467

Change-Id: Id67509d25df92a72eb3683720ad2a95a27b50c89

The media frame cache gets in the way of finding use after free errors of
media frames.  Tools like valgrind and MALLOC_DEBUG don't know when a
frame is released because it gets put into the cache instead of being
freed.

* Added the "cache_media_frames" option to asterisk.conf.  Disabling the
option helps track down media frame mismanagement when using valgrind or
MALLOC_DEBUG.  The cache gets in the way of determining if the frame is
used after free and who freed it.  NOTE: This option has no effect when
Asterisk is compiled with the LOW_MEMORY compile time option enabled
because the cache code does not exist.

To disable the media frame cache simply disable the cache_media_frames
option in asterisk.conf and restart Asterisk.

Sample asterisk.conf setting:
[options]
cache_media_frames=no

ASTERISK-27413

Change-Id: I0ab2ce0f4547cccf2eb214901835c2d951b78c00

This mimics the behavior of Chrome and Firefox and creates an ephemeral
X.509 certificate for each DTLS session.

Currently, the only supported key type is ECDSA because of its faster
generation time, but other key types can be added in the future as
necessary.

ASTERISK-27395

Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4

Fix typo, that specify usage wrong option 'dtmf-features' for CHANNEL() function
instead of correct 'dtmf_features'

ASTERISK-27377 #close

Change-Id: I15ecc829c1035b359584673e12cdb5c9291ac930

When the identify_by option on an endpoint is set to ip it will
only be identified using the res_pjsip_endpoint_identifier_ip module.
This ensures that it is not mistakenly matched using the username of
the From header. To ensure behavior has not changed the default has
been changed to "username,ip" for the identify_by option.

ASTERISK-27206

Change-Id: I2170b86a7f7e221b4f00bf14aa1ef1ac5b050bbd

* Mark the module deprecated.
* Disable the module by default.
* Produce a warning the first time a macro is used.
* Note deprecation related options in app_dial and app_queue.

ASTERISK-27350

Change-Id: I560ea043bacdbc5534a17d97854273d52c2f1bdc

Add action to cancel feature attended transfer with AMI interface

ASTERISK-27215 #close

Change-Id: Iab8a81362b5a1757e2608f70b014ef863200cb42

When "rewrite_contact" is enabled, the "max_contacts" count option can
block re-registrations because the source port from the endpoint can be
random.  When the re-registration is blocked, the endpoint may give up
re-registering and require manual intervention.

* The "remove_existing" option now allows a registration to succeed by
displacing any existing contacts that now exceed the "max_contacts" count.
Any removed contacts are the next to expire.  The behaviour change is
beneficial when "rewrite_contact" is enabled and "max_contacts" is greater
than one.  The removed contact is likely the old contact created by
"rewrite_contact" that the device is refreshing.

ASTERISK-27192

Change-Id: I64c107a10b70db1697d17136051ae6bf22b5314b

A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to
receive unsolicited MWI NOTIFY requests and make them available to
other modules via the stasis message bus.

res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request"
that parses a simple-message-summary body and, if
endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state
with the voice-message counts from the message.

Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c

Add ability to use tokens instead of passwords according to Google OAuth 2.0
protocol.

ASTERISK-27169
Reported by: Andrey Egorov
Tested by: Andrey Egorov

Change-Id: I07f7052a502457ab55010a4d3686653b60f4c8db

'--with-pjproject-bundled' is now the default when running
./configure. It can be disabled with '--without-pjproject-bundled'.

To make building without an internet connection easier, a new
./configure option '--with-download-cache' was added that sets
the cache for externals (like pjproject, the codecs and the DPMA),
AND the sounds files.  It can also be specified as an environment
variable named "AST_DOWNLOAD_CACHE".  The existing
'--with-sounds-cache' option / SOUNDS_CACHE_DIR env variable and
'--with-externals-cache' option / EXTERNALS_CACHE_DIR env variable
remain and if specified, will override '--with-downloads-cache'.

ASTERISK-27189

Change-Id: Ifa9783fddf44aafadb060c9feba713dfa81d38ce

This adds a way to access information passed along with SIP headers in
a REFER message that initiates a transfer. Headers matching a dialplan
variable GET_TRANSFERRER_DATA in the transferrer channel are added to
a HASH object TRANSFER_DATA to be accessed with functions HASHKEY and HASH.

The variable GET_TRANSFERRER_DATA is interpreted to be a prefix for
headers that should be put into the hash. If not set, no headers are
included. If set to a string (perhaps 'X-' in a typical case), all headers
starting this string are added. Empty string matches all headers.

If there are multiple of the same header, only the latest occurrence in
the REFER message is available in the hash.

Obviously, the variable GET_TRANSFERRER_DATA must be inherited by the
referrer channel, and should be set with the '_' or '__' prefix.

I avoided a specific reference to SIP or REFER, as in my mind the mechanism
can be generalized to other channel techs.

ASTERISK-27162

Change-Id: I73d7a1e95981693bc59aa0d5093c074b555f708e

Syntax: SIP_HEADERS([prefix])

If the argument is specified, only the headers matching the given prefix
are returned.

The function returns a comma-separated list of SIP header names from an
incoming INVITE message. Multiple headers with the same name are included
in the list only once. The returned list can be iterated over using the
functions POP() and SIP_HEADER().

For example, '${SIP_HEADERS(Co)}' might return the string
'Contact,Content-Length,Content-Type'.

Practical use is rather '${SIP_HEADERS(X-)}' to enumerate optional
extended headers sent by a peer.

ASTERISK-27163

Change-Id: I2076d3893d03a2f82429f393b5b46db6cf68a267

The "external_media_address" option on transports is now
resolved using dnsmgr. This allows it to be automatically
refreshed regularly if refreshes are enabled in dnsmgr.
If the system is using a dynamic IP address a dynamic DNS
hostname can be provided to keep the IP address up to
date.

Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2

Setting this option will cause the Queue application to only announce
the caller's position if it has improved since the last time that we
announced it.

Change-Id: I173a124121422209485b043e2bf784f54242fce6

This function is a replica of SIPDtmfMode, allowing the DTMF mode of a
PJSIP call to be modified on a per-call basis

ASTERISK-27085 #close

Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612

By default, when res_musiconhold reloads or unloads, it sends a HUP
signal to custom applications (and all descendants), waits 100ms,
then sends a TERM signal, waits 100ms, then finally sends a KILL
signal.  An application which is interacting with an external
device and/or spawns children of its own may not be able to exit
cleanly in the default times, expecially if sent a KILL signal, or
if it's children are getting signals directly from
res_musiconhoild.

* To allow extra time, the 'kill_escalation_delay'
  class option can be used to set the number of milliseconds
  res_musiconhold waits before escalating kill signals, with the
  default being the current 100ms.

* To control to whom the signals are sent, the "kill_method" class
  option can be set to "process_group" (the default, existing
  behavior), which sends signals to the application and its
  descendants directly, or "process" which sends signals only to the
  application itself.

Change-Id: Iff70a1a9405685a9021a68416830c0db5158603b

There wasn't any good way to pass options like --host or --build
down to the pjproject configure which makes cross-compiling difficult.

* Added a new PJPROJECT_CONFIGURE_OPTS environment variable which
  can be used to pass arbitrary options to pjproject configure.
* Automatically set the pjproject configure --host and --build
  options to match those supplied for the asterisk configure.

ASTERISK-27097 #close
Reported-by: Kinsey Moore

Change-Id: I5fa776e110262851173002a26ffe1172e4c35b2e

The existing auto dtmf mode reverts to inband if 4733 fails to be
negotiated.  This patch adds a new mode auto_info which will
switch to INFO instead of inband if 4733 is not available.

ASTERISK-27066 #close

Change-Id: Id185b11e84afd9191a2f269e8443019047765e91

A new global option "imap_poll_logout" was added to specify whether need to
disconnect from the IMAP server after polling of mailboxes.

ASTERISK-27068 #close

Closing IMAP connection after loading mailbox from voicemail.conf

ASTERISK-24052 #close

Change-Id: Ib7558ba04516240a32b65f42e9be64372a0ae12a

This option was added to control whether to notify dialog-info state
'early' or 'confirmed' on Ringing when already INUSE.
The value "yes" is useful for some SIP phones (Cisco SPA)
to be able to indicate and pick up ringing devices.

ASTERISK-26919 #close

Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711