CHANGES

==============================================================================
===
=== This file documents the new and/or enhanced functionality added in
=== the Asterisk versions listed below. This file does NOT include
=== changes in behavior that would not be backwards compatible with
=== previous versions; for that information see the UPGRADE.txt file
=== and the other UPGRADE files for older releases.
===
==============================================================================

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.18.0 to Asterisk 13.19.0 ----------
------------------------------------------------------------------------------

Core
------------------
 * 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.

chan_sip
------------------
 * Calls to invalid extensions are now reported as an ACL failure security event
   "no_extension_match".

res_pjsip
------------------
 * The "identify_by" on endpoints can now be set to "ip" to restrict an endpoint
   being matched based only on IP address. To ensure no behavior change the
   default has been changed to "username,ip".

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.17.0 to Asterisk 13.18.0 ----------
------------------------------------------------------------------------------

Core
------------------
 * VP9 is now a supported passthrough video codec and it can be used by
   specifying "vp9" in the allow line.

Build System
------------------
 * A '--with-download-cache' option is now available which is equivalent to
   setting '--with-sounds-cache' and '--with-externals-cache' to the same
   value.  The download cache can also be set via the AST_DOWNLOAD_CACHE
   environment variable.

res_pjsip
------------------
 * The "external_media_address" on transports is now resolved using dnsmgr and
   when dnsmgr refreshes are enabled will be automatically updated with the new
   IP address of a given hostname.

 * 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.

 * 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.

res_musiconhold
------------------
 * 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.

res_pjsip
------------------
 * New dialplan function PJSIP_DTMF_MODE added to get or change the DTMF mode
   of a channel on a per-call basis.

res_xmpp
-----------------
 * OAuth 2.0 authentication is now supported when contacting Google. Follow the
   instructions in xmpp.conf.sample to retrieve and configure the necessary
   tokens.

app_queue
------------------
 * Add priority to callers in AMI QueueStatus response.

AMI
------------------
 * Added a new CancelAtxfer action that cancels an attended transfer.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.16.0 to Asterisk 13.17.0 ----------
------------------------------------------------------------------------------

app_voicemail
------------------
 * A new global option "imap_poll_logout" was added to specify whether need to
   disconnect from the IMAP server after polling of mailboxes.
   Default: no

res_pjsip
------------------
 * A new endpoint option "refer_blind_progress" was added to turn off notifying
   the progress details on Blind Transfer. If this option is not set then
   the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted".
   On default is enabled.
   Some SIP phones like Mitel/Aastra or Snom keep the line busy until
   receive "200 OK".

 * A new endpoint option "notify_early_inuse_ringing" was added to control
   whether to notify dialog-info state 'early' or 'confirmed' on Ringing
   when already INUSE.

 * The endpoint option 'dtmf_mode' has a new option 'auto_dtmf' added. This
   mode works similar to 'auto' except uses DTMF INFO as fallback instead of
   INBAND.

res_agi
------------------
 * The EAGI() application will now look for a dialplan variable named
   EAGI_AUDIO_FORMAT and use that format with the 'enhanced' audio pipe that
   EAGI provides. If not specified, it will continue to use the default signed
   linear (slin).

chan_pjsip
------------------
 * When dialing an endpoint directly or using the PJSIP_DIAL_CONTACTS dialplan
   function any contact which is considered unreachable due to qualify being
   enabled will no longer be called.

 * The asymmetric_rtp_codec option now also controls whether chan_pjsip will
   send media as-is without transcoding if the codec has been negotiated in the
   SDP. If set to "no" then Asterisk will only ever send the preferred codec
   from the SDP, unless the remote side sends a different codec and we will
   switch to match.

Build System
------------------
 * Added a new PJPROJECT_CONFIGURE_OPTS environment variable which can be used
   to pass arbitrary options to the bundled pjproject configure.

 * Automatically set the bundled pjproject configure --host and --build
   options to match those supplied for the asterisk configure.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.15.0 to Asterisk 13.16.0 ----------
------------------------------------------------------------------------------

res_rtp_asterisk
------------------
 * Added the stun_blacklist option to rtp.conf.  Some multihomed servers have
   IP interfaces that cannot reach the STUN server specified by stunaddr.
   Blacklist those interface subnets from trying to send a STUN packet to find
   the external IP address.  Attempting to send the STUN packet needlessly
   delays processing incoming and outgoing SIP INVITEs because we will wait
   for a response that can never come until we give up on the response.
   Multiple subnets may be listed.

Logging
-------------------
 * Added logger_queue_limit to the configuration options.
   All log messages go to a queue serviced by a single thread
   which does all the IO.  This setting controls how big that
   queue can get (and therefore how much memory is allocated)
   before new messages are discarded.
   The default is 1000.

res_pjsip_config_wizard
------------------
 * Two new parameters have been added to the pjsip config wizard.
   Setting 'sends_line_with_registrations' to true will cause the wizard
   to skip the creation of an identify object to match incoming requests
   to the endpoint and instead add the line and endpoint parameters to
   the outbound registration object.
   Setting 'outbound_proxy' is a shortcut for adding individual
   endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy
   parameters.

res_hep_rtcp
------------------
 * If the 'call-id' value is specified for the uuid_type option and a
   chan_sip channel is used the resulting HEP traffic will now contain the
   SIP Call-ID instead of the Asterisk channel name.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.14.0 to Asterisk 13.15.0 ----------
------------------------------------------------------------------------------

AMI
------------------
 * The 'PJSIPShowEndpoint' command's respone event of 'IdentifyDetail' now
   contains a new optional parameter, 'MatchHeader', mapping to the new
   configuration option 'match_header' for the corresponding 'identify' object.
   It should be noted that since 'match_header' takes in a key: value pair, the
   event parameter will contain a ':' as well.

app_record
------------------
 * Added new 'u' option to Record() application which prevents Asterisk from
   truncating silence from the end of recorded files.

res_pjsip_outbound_registration
------------------
 * Outbound registrations are now refreshed when res_stun_monitor detects
   a network change event has happened.
   The 'pjsip send (un)register' CLI commands were updated to accept '*all'
   as an argument to operate on all registrations.
   The 'PJSIP(Un)Register' AMI commands were updated to also accept '*all'.

app_voicemail
------------------
 * The 'Comedian Mail' prompts can now be overriden using the 'vm-login' and
   'vm-newuser' configuration options in voicemail.conf.

 * Added 'fromstring' field to the voicemail boxes. If set, it will override
   the global 'fromstring' field on a per-mailbox basis.

func_channel
------------------
 * Added CHANNEL(callid) to retrieve the call log tag associated with the
   channel.  e.g., [C-00000000]  Dialplan now has access to the call log
   search key associated with the channel so it can be saved in case there
   is a problem with the call.

res_pjsip
------------------
 * A new transport parameter 'symmetric_transport' has been added.
   When a request from a dynamic contact comes in on a transport with this
   option set to 'yes', the transport name will be saved and used for
   subsequent outgoing requests like OPTIONS, NOTIFY and INVITE.  It's
   saved as a contact uri parameter named 'x-ast-txp' and will display with
   the contact uri in CLI, AMI, and ARI output.  On the outgoing request,
   if a transport wasn't explicitly set on the endpoint AND the request URI
   is not a hostname, the saved transport will be used and the 'x-ast-txp'
   parameter stripped from the outgoing packet.  To facilitate recreation of
   subscriptions on asterisk restart, a new column 'contact_uri' needed to be
   added to the ps_subcsription_persistence table.  Since new columns were
   added to both transport and subscription_persistence, an alembic upgrade
   should be run to bring the database tables up to date.

 * A new option, allow_overlap, has been added to endpoints which allows
   overlap dialing functionality to be enabled or disabled. The option defaults
   to enabled.

res_pjsip_transport_websocket
------------------
 * Removed non-secure websocket support.  Firefox and Chrome have not allowed
   non-secure websockets for quite some time so this shouldn't be an issue
   for people.  Attempting to use a non-secure websocket may or may not work
   when Asterisk attempts to send SIP requests to do something like initiate
   call hangup.

res_pjsip_endpoint_identifier_ip
------------------
 * A new option has been added to the 'identify' configuration object,
   'match_header'. The 'match_header' attribute should contain a SIP
   header: value pair that, When set, will cause inbound requests that contain
   the matching SIP header/value pair to be associated with the corresponding
   endpoint. This option is cumulative with the 'match' option, so that if
   either option matches the request, the request is associated with the
   endpoint.

   In a future release, this module will be renamed to something more
   appropriate, as it now matches inbound requests on more than just IP
   address.

res_rtp_asterisk
-----------------
 * The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
   Data and Control Packets on a Single Port." So far, the only channel driver
   that supports this feature is chan_pjsip. You can set "rtcp_mux = yes" on
   a PJSIP endpoint in pjsip.conf to enable the feature.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.13.0 to Asterisk 13.14.0 ----------
------------------------------------------------------------------------------

res_pjproject
------------------
 * Added new CLI command "pjproject set log level".  The new command allows
   the maximum PJPROJECT log levels to be adjusted dynamically and
   independently from the set debug logging level like many other similar
   module debug logging commands.

 * Added new companion CLI command "pjproject show log level" to allow the
   user to see the current maximum pjproject logging level.

 * Added new pjproject.conf startup section "log_level' option to set the
   initial maximum PJPROJECT logging level.

res_pjsip_outbound_registration
------------------
 * Statsd no longer logs redundant status PJSIP.registrations.state changes
   for internal state transitions that don't change the reported public status
   state.

res_pjsip_registrar
------------------
 * The PJSIPShowRegistrationInboundContactStatuses AMI command has been added
   to return ContactStatusDetail events as opposed to
   PJSIPShowRegistrationsInbound which just a dumps every defined AOR.

res_pjsip
------------------
 * Six existing contact fields have been added to the end of the
   ContactStatusDetail AMI event:
   ID, AuthenticateQualify, OutboundProxy, Path, QualifyFrequency and
   QualifyTimeout.  Existing fields have not been disturbed.

res_pjsip_endpoint_identifier_ip
------------------
 * SRV lookups can now be done on provided hostnames to determine additional
   source IP addresses for requests. This is configurable using the
   "srv_lookups" option on the identify and defaults to "yes".

ARI
------------------
 * The 'ari set debug' command has been enhanced to accept 'all' as an
   application name.  This allows dumping of all apps even if an app
   hasn't registered yet.

 * 'ari set debug' now displays requests and responses as well as events.
 
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.12.0 to Asterisk 13.13.0 ----------
------------------------------------------------------------------------------

AMI
------------------
 * Events that reference a bridge may now contain two new optional fields:
   - 'BridgeVideoSourceMode': the video source mode for the bridge.
     Can be one of 'none', 'talker', or 'single'.
   - 'BridgeVideoSource': the unique ID of the channel that is the video
     source in this bridge, if one exists.

 * A new event, BridgeVideoSourceUpdate, has been added with a class
   authorization of CALL. The event is raised when the video source changes
   in a multi-party mixing bridge.

ARI
------------------
 * The bridges resource now exposes two new operations:
   - POST /bridges/{bridgeId}/videoSource/{channelId}: Set a video source in a
     multi-party mixing bridge
   - DELETE /bridges/{bridgeId}/videoSource: Remove the set video source,
     reverting to talk detection for the video source

 * The bridge model in any returned response or event now contains the following
   optional fields:
   - video_mode: the video source mode for the bridge. Can be one of 'none',
     'talker', or 'single'.
   - video_source_id: the unique ID of the channel that is the video source
     in this bridge, if one exists.

 * A new event, BridgeVideoSourceChanged, has been added for bridges.
   Applications subscribed to a bridge will receive this event when the source
   of video changes in a mixing bridge.

res_pjsip
------------------
 * Automatic dual stack support is now implemented. Depending on DNS resolution
   and the transport used for sending a message the SIP signaling and SDP will
   be updated with the correct IP address and protocol version. This means that
   the rtp_ipv6 and t38_udptl_ipv6 options no longer have any effect. The
   res_pjsip_multihomed module has also been moved into core res_pjsip to ensure
   that messages are updated with the correct address information in all cases.

chan_pjsip
------------------
 * The default behavior for RTP codecs has been changed. The sending codec will
   now match the receiving codec. This can be turned off and behavior reverted
   to asymmetric using the "asymmetric_rtp_codec" endpoint option. If this
   option is set then the sending and received codec are allowed to differ.

CLI Commands
------------------
 * Three new CLI commands have been added for ARI:
   - ari show apps:
      Displays a listing of all registered ARI applications.
   - ari show app <name>:
      Display detailed information about a registered ARI application.
   - ari set debug <name> <on|off>:
      Enable/disable debugging of an ARI application. When debugged, verbose
      information will be sent to the Asterisk CLI.

RTP
------------------
 * New setting "rtp_pt_dynamic = 96" in asterisk.conf:
   Normally the Dynamic RTP Payload Type numbers are 96-127, which allow 32
   formats. When you use more and receive the message "No Dynamic RTP mapping
   available", extend the dynamic range by going for rtp_pt_dynamic = 35 (or 0)
   instead of 96. This allows 29 (or 64) additional formats. On default this is
   disabled and the range is 96-127 because any number below might be rejected
   by a remote implementation; although no such broken implementation is known.

Queue
------------------
 * A new dialplan variable, ABANDONED, is set when the call is not answered
   by an agent.

Core
------------------
 * The TLS core in Asterisk now supports X.509 certificate subject alternative
   names. This way one X.509 certificate can be used for hosts that can be
   reached under multiple DNS names or for multiple hosts.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.11.0 to Asterisk 13.12.0 ----------
------------------------------------------------------------------------------

Build System
------------------
 * The res_digium_phone, codec_g729a, codec_silk, codec_siren7 and
   codec_siren14 binary modules hosted at downloads.digium.com can now be
   automatically downloaded and installed during the Asterisk install
   process.  If selected in menuselect, when 'make install' is run, the
   script will check the downloads site for a new version and download
   and install it if needed.  The '--with-externals-cache' option to
   ./configure can be used to specify a location to cache the latest
   tarballs so they don't have to be re-downloaded for every install.

app_voicemail
------------------
 * Added "tps_queue_high" and "tps_queue_low" options.
   The options can modify the taskprocessor alert levels for this module.
   Additional information can be found in the sample configuration file at
   config/samples/voicemail.conf.sample.

res_pjsip_mwi
------------------
 * Added "mwi_tps_queue_high" and "mwi_tps_queue_low" global configuration
   options to tune taskprocessor alert levels.

 * Added "mwi_disable_initial_unsolicited" global configuration option
   to disable sending unsolicited MWI to all endpoints on startup.
   Additional information can be found in the sample configuration file at
   config/samples/pjsip.conf.sample.

chan_pjsip
------------------
 * A new dialplan function, PJSIP_SEND_SESSION_REFRESH, has been added. When
   invoked, a re-INVITE or UPDATE request will be sent immediately to the
   endpoint underlying the channel. When used in combination with the existing
   dialplan function PJSIP_MEDIA_OFFER, this allows the formats on a PJSIP
   channel to be re-negotiated and updated after session set up.

res_pjsip
------------------
 * A new endpoint configuration parameter 'contact_user' has been added which
   when set will override the default user set on Contact headers in outgoing
   requests.

 * If you are using a sorcery realtime backend to store global res_pjsip
   options (ps_globals table) then you now have to do a res_pjsip reload for
   changes to these options to take effect.  If you are using pjsip.conf to
   configure these options then you already had to do a reload after making
   changes.

 * Added "ignore_uri_user_options" global configuration option for
   compatibility with an ITSP that sends URI user field options.  When enabled
   the user field is truncated at the first semicolon.
   Example:
   URI: "sip:1235557890;phone-context=national@x.x.x.x;user=phone"
   The user field is "1235557890;phone-context=national"
   Which is truncated to this: "1235557890"

   Note: The caller-id and redirecting number strings obtained from incoming
   SIP URI user fields are now always truncated at the first semicolon.

res_rtp_asterisk
------------------
  * An option, ice_blacklist, has been added which allows certain subnets to be
    excluded from local ICE candidates.

app_confbridge
------------------
  * Some sounds played into the bridge are played asynchronously. This, for
    instance, allows a channel to immediately exit the ConfBridge without having
    to wait for a leave announcement to play.

Core
------------------
  * If Asterisk is built with systemd support, and run under systemd, it will
    notify systemd of its state using sd_notify. Use 'Type=notify' in
    asterisk.service.

app_dial
------------------
 * Added the "Q" option which sets the Q.850/Q.931 cause on unanswered channels
   when another channel answers the call.  The default of ANSWERED_ELSEWHERE
   is unchanged.

res_ari
------------------
 * ARI events will all now include a new field in the root of the JSON message,
   'asterisk_id'.  This will be the unique ID for the Asterisk system
   transmitting the event.  The value can be overridden using the 'entityid'
   setting in asterisk.conf.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.10.0 to Asterisk 13.11.0 ----------
------------------------------------------------------------------------------

chan_dahdi
------------------
 * Added "faxdetect_timeout" option.
   The option determines how many seconds into a call before faxdetect
   is disabled for the call.  Setting the value to zero disables the timeout.

chan_sip
------------------
 * Session-Timers (RFC 4028) work for TCP (and TLS) transports as well now.
   Previously Asterisk dropped calls only with UDP transports. However with
   longer international calls via TCP, the SIP channel might break, because
   all hops on the Internet route must stay online (have not a single power
   outage, for example). Therefore with Session-Timers enabled (which are
   enabled at default), you might see additional dropped calls. Consequently
   please, consider to go for session-timers=refuse in your sip.conf.

res_pjsip
------------------
 * Added "fax_detect_timeout" to endpoint.
   The option determines how many seconds into a call before fax_detect
   is disabled for the call.  Setting the value to zero disables the timeout.

 * Added "subscribe_context" to endpoint.
   If specified, incoming SUBSCRIBE requests will be searched for the matching
   extension in the indicated context. If no "subscribe_context" is specified,
   then the "context" setting is used.

res_rtp_asterisk
------------------
 * The DTLS part in Asterisk now supports Perfect Forward Secrecy (PFS).
   Enabling PFS is attempted by default, and is dependent on the configuration
   of the module using TLS.
   - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
     specify a ECDHE cipher suite in sip.conf, for example:
       dtlscipher=AES128-SHA
   - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
     into the private key file, e.g., sip.conf dtlsprivatekey. For example:
       openssl dhparam -out ./dh.pem 2048
   - Because clients expect the server to prefer PFS, and because OpenSSL sorts
     its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
     Consider re-ordering your cipher suites in the respective configuration
     file. For example:
       dtlscipher=ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256
     which forces PFS and requires at least DTLS 1.2.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.9.0 to Asterisk 13.10.0 -----------
------------------------------------------------------------------------------

Core
------------------
 * A channel variable FORWARDERNAME is now set which indicates which channel
   was responsible for a forwarding requests received on dial attempt.

chan_multicast_rtp
------------------
 * Deprecated in favor of chan_rtp which is basically chan_multicast_rtp
   renamed to chan_rtp with UnicastRTP channels added and some internal code
   improvements.

chan_rtp
------------------
 * The format for dialing a unicast RTP channel is:
   UnicastRTP/<destination-addr>[/[<options>]]
   Where <destination-addr> is something like '127.0.0.1:5060'.
   Where <options> are in standard Asterisk flag options format:
   c(<codec>) - Specify which codec/format to use such as 'ulaw'.
   e(<engine>) - Specify which RTP engine to use such as 'asterisk'.

 * More options are available over what chan_multicast_rtp supports.
   The format for dialing a multicast RTP channel is:
   MulticastRTP/<type>/<destination-addr>[/[<control-addr>][/[<options>]]]
   Where <type> can be either 'basic' or 'linksys'.
   Where <destination-addr> is something like '224.0.0.3:5060'.
   Where <control-addr> is something like '127.0.0.1:5060'.
   Where <options> are in standard Asterisk flag options format:
   c(<codec>) - Specify which codec/format to use such as 'ulaw'.
   i(<address>) - Specify the interface address from which multicast RTP
     is sent.
   l(<enable>) - Set whether packets are looped back to the sender.  The
     enable value can be 0 to set looping to off and non-zero to set
     looping on.
   t(<ttl>) - Set the time-to-live (TTL) value for multicast packets.

func_odbc
------------------
 * Added new global option "single_db_connection".
   Enabling this option func_odbc will use a single database connection per DSN.
   This option is enabled by default.

res_fax
------------------
 * Added FAXMODE variable to let dialplan know what fax transport was used.
   FAXMODE variable is set to either "audio" or "T38".

res_pjsip
------------------
 * Added "via_addr", "via_port", "call_id" to contacts.
   As res_pjsip_nat rewrites contact's address, only the last Via header
   can contain the source address of registered endpoint.
   Also Call-Id header may contain the source address of registered endpoint.
   Added new fields ViaAddress,CallID to AMI event ContactStatus

 * Endpoint IP Access Controls
   Added new configuration Endpoint options:
    "acl" - list of IP ACL section names in acl.conf
    "deny" - List of IP addresses to deny access from
    "permit" - List of IP addresses to permit access from
    "contact_acl" - List of Contact ACL section names in acl.conf
    "contact_deny" - List of Contact header addresses to deny
    "contact_permit" - List of Contact header addresses to permit

 * Added "reg_server" to contacts.
   If the Asterisk system name is set in asterisk.conf, it will be stored
   into the "reg_server" field in the ps_contacts table to facilitate
   multi-server setups.

 * When starting Asterisk, received traffic will now be ignored until Asterisk
   has loaded all modules and is fully booted.

res_hep
------------------
 * Added a new option, 'uuid_type', that sets the preferred source of the Homer
   correlation UUID. The valid options are:
   - call-id: Use the PJSIP SIP Call-ID header value
   - channel: Use the Asterisk channel name
   The default value is 'call-id'. In the event that a HEP module cannot find a
   valid value using the specified 'uuid_type', the module may fallback to a
   more readily available source for the correlation UUID.

res_pjsip_info_empty
--------------------
 * A new module that can respond to empty Content-Type INFO packets during call.
   Some SBCs will terminate a call if their empty INFO packets are not responded
   to within a predefined time.

res_odbc
------------------
 * A new option has been added, 'max_connections', which sets the maximum number
   of concurrent connections to the database. This option defaults to 1 which
   returns the behavior to that of Asterisk 13.7 and prior.

app_confbridge
------------------
 * Added a bridge profile option called regcontext that allows you to
   dynamically register the conference bridge name as an extension into
   the specified context.  This allows tracking down conferences on multi-
   server installations via alternate means (DUNDI for example). By default
   this feature is not used.

Codecs
------------------
 * Added the associated format name to 'core show codecs'.

res_ari_channels
------------------
 * Added 'formats' to channel create/originate to allow setting the allowed
   formats for a channel when no originator channel is available.  Especially
   useful for Local channel creation where no other format information is
   available.  'core show codecs' can now be used to look up suitable format
   names.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.8.0 to Asterisk 13.9.0 ------------
------------------------------------------------------------------------------

res_pjsip
------------------
 * Added new global option (disable_multi_domain) to pjsip.
   Disabling Multi Domain can improve realtime performace by reducing
   number of database requsts.

chan_pjsip
------------------
 * Added 'pjsip show channelstats' CLI command.

res_pjsip_outbound_publish
------------------
 * Added support for setting the transport used on outbound publish
   using the transport configuration option.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.7.0 to Asterisk 13.8.0 ------------
------------------------------------------------------------------------------

res_pjsip_caller_id
------------------
 * Per RFC3325, the 'From' header is now anonymized on outgoing calls when
   caller id presentation is prohibited.

res_pjsip_config_wizard
------------------
 * A new command (pjsip export config_wizard primitives) has been added that
   will export all the pjsip objects it created to the console or a file
   suitable for reuse in a pjsip.conf file.

Build System
------------------
 * To help insure that Asterisk is compiled and run with the same known
   version of pjproject, a new option (--with-pjproject-bundled) has been
   added to ./configure.  When specified, the version of pjproject specified
   in third-party/versions.mak will be downloaded and configured.  When you
   make Asterisk, the build process will also automatically build pjproject
   and Asterisk will be statically linked to it.  Once a particular version
   of pjproject is configured and built, it won't be configured or built
   again unless you run a 'make distclean'.

   To facilitate testing, when 'make install' is run, the pjsua and pjsystest
   utilities and the pjproject python bindings will be installed in
   ASTDATADIR/third-party/pjproject.

   The default behavior remains building with the shared pjproject
   installation, if any.

app_confbridge
------------------
 * Added CONFBRIDGE_INFO(muted,) for querying the muted conference state.

 * Added Muted header to AMI ConfbridgeListRooms action response list events
   to indicate the muted conference state.

 * Added Muted column to CLI "confbridge list" output to indicate the muted
   conference state and made the locked column a yes/no value instead of a
   locked/unlocked value.

REDIRECTING(reason)
------------------
 * The REDIRECTING(reason) value is now treated consistently between
   chan_sip and chan_pjsip.

   Both channel drivers match incoming reason values with values documented
   by REDIRECTING(reason) and values documented by RFC5806 regardless of
   whether they are quoted or not.  RFC5806 values are mapped to the
   equivalent REDIRECTING(reason) documented value and is set in
   REDIRECTING(reason).  e.g., an incoming RFC5806 'unconditional' value or a
   quoted string version ('"unconditional"') is converted to
   REDIRECTING(reason)'s 'cfu' value.  The user's dialplan only needs to deal
   with 'cfu' instead of any of the aliases.

   The incoming 480 response reason text supported by chan_sip checks for
   known reason values and if not matched then puts quotes around the reason
   string and assigns that to REDIRECTING(reason).

   Both channel drivers send outgoing known REDIRECTING(reason) values as the
   unquoted RFC5806 equivalent.  User custom values are either sent as is or
   with added quotes if SIP doesn't allow a character within the value as
   part of a RFC3261 Section 25.1 token.  Note that there are still
   limitations on what characters can be put in a custom user value.  e.g.,
   embedding quotes in the middle of the reason string is just going to cause
   you grief.

 * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases.
   e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the
   'cfu' value.

res_pjproject
------------------
 * This module is the successor of res_pjsip_log_forwarder.  As well as
   handling the log forwarding (which now displays as 'pjproject:0' instead
   of 'pjsip:0'), it also adds a 'pjproject show buildopts' command to the CLI.
   This displays the compiled-in options of the pjproject installation
   Asterisk is currently running against.

 * Another feature of this module is the ability to map pjproject log levels
   to Asterisk log levels, or to suppress the pjproject log messages
   altogether.  Many of the messages emitted by pjproject itself are the result
   of errors which Asterisk will ultimately handle so the messages can be
   misleading or just noise.  A new config file (pjproject.conf) has been added
   to configure the mapping and a new CLI command (pjproject show log mappings)
   has been added to display the mappings currently in use.

res_pjsip
------------------
 * Transports are now reloadable.  In testing, no in-progress calls were
   disrupted if the ip address or port weren't changed, but the possibility
   still exists.  To make sure there are no unintentional drops, a new option
   'allow_reload', which defaults to 'no' has been added to transport.  If
   left at the default, changes to the particular transport will be ignored.
   If set to 'yes', changes (if any) will be applied.

 * Added new global option (regcontext) to pjsip. When set, Asterisk will
   dynamically create and destroy a NoOp priority 1 extension
   for a given endpoint who registers or unregisters with us.

 * Endpoints and aors can now be identified by the username and realm in an
   incoming Authorization header.  To use this feature, add "auth_username"
   to your endpoint's "identify_by" list.  You can combine "auth_username"
   and the original "username" to test both the From/To and Authorization
   headers.  For endpoints, the order is controlled by the global
   "endpoint_identifier_order" setting.  For matching aors to an endpoint
   for inbound registration, the order is controlled by this option.

 * In conjunction with the "auth_username" change, 3 new options have been
   added to the global configuration object that control how many unidentified
   requests over a certain period from the same IP address can be received
   before a security altert is generated.  A new CLI command
   "pjsip show unidentified_requests" will list the current candidates.

res_pjsip_history
------------------
 * A new module, res_pjsip_history, has been added that provides SIP history
   viewing/filtering from the CLI. The module is intended to be used on systems
   with busy SIP traffic, where existing forms of viewing SIP messages - such
   as the res_pjsip_logger - may be inadequate. The module provides two new
   CLI commands:
   - 'pjsip set history {on|off|clear}' - this enables/disables SIP history
     capturing, as well as clears an existing history capture. Note that SIP
     packets captured are stored in memory until cleared. As a result, the
     history capture should only be used for debugging/viewing purposes, and
     should *NOT* be left permanently enabled on a system.
   - 'pjsip show history' - displays the captured SIP history. When invoked
     with no options, the entire captured history is displayed. Two options
     are available:
     -- 'entry <num>' - display a detailed view of a single SIP message in
        the history
     -- 'where ...' - filter the history based on some expression. For more
        information on filtering, view the current CLI help for the
        'pjsip show history' command.

Voicemail
------------------
 * app_voicemail and res_mwi_external can now be built together.  The default
   remains to build app_voicemail and not res_mwi_external but if they are
   both built, the load order will cause res_mwi_external to load first and
   app_voicemail will be skipped.  Use 'preload=app_voicemail.so' in
   modules.conf to force app_voicemail to be the voicemail provider.

Queue
-------------------
 * Added field ReasonPause on QueueMemberStatus if set when paused, the reason
   the queue member was paused.

res_pjsip_sdp_rtp
------------------
 * A new option (bind_rtp_to_media_address) has been added to endpoint which
   will cause res_pjsip_sdp_rtp to actually bind the RTP instance to the
   media_address as well as using it in the SDP.  If set, RTP packets will now
   originate from the media address instead of the operating system's "primary"
   ip address.

res_rtp_asterisk
------------------
 * A new configuration section - ice_host_candidates - has been added to
   rtp.conf, allowing automatically discovered ICE host candidates to be
   overriden. This allows an Asterisk server behind a 1:1 NAT to send its
   external IP as a host candidate rather than relying on STUN to discover it.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.6.0 to Asterisk 13.7.0 ------------
------------------------------------------------------------------------------

Codecs
------------------
 * Added format attribute negotiation for the VP8 video codec. Format attribute
   negotiation is provided by the res_format_attr_vp8 module.

ConfBridge
------------------
 * A new "timeout" user profile option has been added. This configures the number
   of seconds that a participant may stay in the ConfBridge after joining. When
   the time expires, the user is ejected from the conference and CONFBRIDGE_RESULT
   is set to "TIMEOUT" on the channel.

chan_sip
------------------
 * The websockets_enabled option has been added to the general section of
   sip.conf.  The option is enabled by default to match the previous behavior.
   The option should be disabled when using res_pjsip_transport_websockets to
   ensure chan_sip will not conflict with PJSIP websockets.

Dialplan Functions
------------------
 * The HOLD_INTERCEPT dialplan function now actually exists in the source tree.
   While support for the events was added in Asterisk 13.4.0, the function
   accidentally never made it in. That function is now present, and will cause
   the 'hold' raised by a channel to be intercepted and converted into an
   event instead.

res_pjsip_outbound_registration
-------------------------------
 * A new 'fatal_retry_interval' option has been added to outbound registration.
   When set (default is zero), and upon receiving a failure response to an
   outbound registration, registration is retried at the given interval up to
   'max_retries'.

 * If res_statsd is loaded and a StatsD server is configured, basic statistics
   regarding the state of outbound registrations will now be emitted. This
   includes:
   - A GAUGE statistic for the overall number of outbound registrations, i.e.:
       PJSIP.registrations.count
   - A GAUGE statistic for the overall number of outbound registrations in a
     particular state, e.g.:
       PJSIP.registrations.state.Registered

res_pjsip
------------------
 * The ability to use "like" has been added to the pjsip list and show
   CLI commands.  For instance: CLI> pjsip list endpoints like abc

 * If res_statsd is loaded and a StatsD server is configured, basic statistics
   regarding the state of PJSIP contacts will now be emitted. This includes:
   - A GAUGE statistic for the overall number of contacts in a particular
     state, e.g.:
       PJSIP.contacts.states.Reachable
   - A TIMER statistic for the RTT time for each qualified contact, e.g.:
       PJSIP.contacts.alice@@127.0.0.1:5061.rtt

res_sorcery_memory_cache
------------------------
 * A new caching strategy, full_backend_cache, has been added which caches
   all stored objects in the backend. When enabled all objects will be
   expired or go stale according to the configuration. As well when enabled
   all retrieval operations will be performed against the cache instead of
   the backend.

func_callerid
-------------------
 * CALLERID(pres) is now documented as a valid alternative to setting both
   CALLERID(name-pres) and CALLERID(num-pres) at once.  Some channel drivers,
   like chan_sip, don't make a distinction between the two: they take the
   least public value from name-pres and num-pres.  By using CALLERID(pres)
   for reading and writing, you touch the same combined value in the dialplan.
   The same applies to CONNECTEDLINE(pres), REDIRECTING(orig-pres),
   REDIRECTING(to-pres) and REDIRECTING(from-pres).

res_endpoint_stats
-------------------
 * A new module that emits StatsD statistics regarding Asterisk endpoints.
   This includes a total count of the number of endpoints, the count of the
   number of endpoints in the technology agnostic state of the endpoint -
   online or offline - as well as the number of channels associated with each
   endpoint. These are recorded as three different GAUGE statistics:
    - endpoints.count
    - endpoints.state.{unknown|offline|online}
    - endpoints.{tech}.{resource}.channels


------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.5.0 to Asterisk 13.6.0 ------------
------------------------------------------------------------------------------

Dialplan Functions
------------------
 * The CHANNEL function, when used on a PJSIP channel, now exposes a 'call-id'
   extraction option when using with the 'pjsip' signalling option. It will
   return the SIP Call-ID associated with the INVITE request that established
   the PJSIP channel.

ARI
------------------
 * Two new endpoint related events are now available: PeerStatusChange and
   ContactStatusChange. In particular, these events are useful when subscribing
   to all event sources, as they provide additional endpoint related
   information beyond the addition/removal of channels from an endpoint.
   The ARI version has been bumped to 1.9.0 as a result.

 * Added the ability to subscribe to all ARI events in Asterisk, regardless
   of whether the application 'controls' the resource. This is useful for
   scenarios where an ARI application merely wants to observe the system,
   as opposed to control it. There are two ways to accomplish this:
   (1) Via the WebSocket connection URI. A new query paramter, 'subscribeAll',
       has been added that, when present and True, will subscribe all
       specified applications to all ARI event sources in Asterisk.
   (2) Via the applications resource. An ARI client can, at any time, subscribe
       to all resources in an event source merely by not providing an explicit
       resource. For example, subscribing to an event source of 'channels:'
       as opposed to 'channels:12345' will subscribe the application to all
       channels.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.4.0 to Asterisk 13.5.0 ------------
------------------------------------------------------------------------------

AMI
------------------
 * A new ContactStatus event has been added that reflects res_pjsip contact
   lifecycle changes:  Created, Removed, Reachable, Unreachable, Unknown.

 * Added the Linkedid header to the common channel headers listed for each
   channel in AMI events.

ARI
------------------
 * A new feature has been added that enables the retrieval of modules and
   module information through an HTTP request. Information on a single module