==============================================================================

===

=== 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 16 to Asterisk 17 --------------------

------------------------------------------------------------------------------

chan_sip

------------------

* The chan_sip module is now deprecated, users should migrate to the

replacement module chan_pjsip. See guides at the Asterisk Wiki:

https://wiki.asterisk.org/wiki/x/tAHOAQ

https://wiki.asterisk.org/wiki/x/hYCLAQ

Channels

------------------

* The core no longer uses the stasis cache for channels snapshots.

The following APIs are no longer available:

ast_channel_topic_cached()

ast_channel_topic_all_cached()

The ast_channel_cache_all() and ast_channel_cache_by_name() functions

now returns an ao2_container of ast_channel_snapshots rather than a

container of stasis_messages therefore you can't call stasis_cache

functions on it.

The ast_channel_topic_all() function now returns a normal topic,

not a cached one so you can't use stasis cache functions on it either.

The ast_channel_snapshot_type() stasis message now has the

ast_channel_snapshot_update structure as it's data.

ast_channel_snapshot_get_latest() still returns the latest snapshot.

Bridging

------------------

* The bridging core no longer uses the stasis cache for bridge

snapshots. The latest bridge snapshot is now stored on the

ast_bridge structure itself.

* The following APIs are no longer available since the stasis cache

is no longer used:

ast_bridge_topic_cached()

ast_bridge_topic_all_cached()

* A topic pool is now used for individual bridge topics.

* The ast_bridge_cache() function was removed since there's no

longer a separate container of snapshots.

* A new function "ast_bridges()" was created to retrieve the

container of all bridges. Users formerly calling

ast_bridge_cache() can use the new function to iterate over

bridges and retrieve the latest snapshot directly from the

bridge.

* The ast_bridge_snapshot_get_latest() function was renamed to

ast_bridge_get_snapshot_by_uniqueid().

* A new function "ast_bridge_get_snapshot()" was created to retrieve

the bridge snapshot directly from the bridge structure.

* The ast_bridge_topic_all() function now returns a normal topic

not a cached one so you can't use stasis cache functions on it

either.

* The ast_bridge_snapshot_type() stasis message now has the

ast_bridge_snapshot_update structure as it's data. It contains

the last snapshot and the new one.

------------------------------------------------------------------------------

--- Functionality changes from Asterisk 16.1.0 to Asterisk 16.2.0 ------------

------------------------------------------------------------------------------

res_pjsip

------------------

* Added "send_contact_status_on_update_registration" global configuration option

to enable sending AMI ContactStatus event when a device refreshes its registration.

Features

------------------

* Before Asterisk 12, when using the automon or automixmon features defined

in features.conf, a channel variable (TOUCH_MIXMONITOR_OUTPUT) was set on

both channels, indicating the filename of the recording.

When bridging was overhauled in Asterisk 12, the behavior was changed such

that the variable was only set on the peer channel and not on the channel

that initiated the automon or automixmon.

The previous behavior has been restored so both channels receive the

channel variable when one of these features is invoked.

------------------------------------------------------------------------------

--- Functionality changes from Asterisk 16.0.0 to Asterisk 16.1.0 ------------

------------------------------------------------------------------------------

pbx_config

------------------

* pbx_config will now find and process multiple 'globals' sections from

extensions.conf. Variables are processed in the order they are found

and duplicate variables overwrite the previous value.

chan_pjsip

------------------

* New dialplan function PJSIP_PARSE_URI added to parse an URI and return

a specified part of the URI.

Core

------------------

* ast_bt_get_symbols() now returns a vector of strings instead of an

array of strings. This must be freed with ast_bt_free_symbols.

res_pjsip

------------------

* New options 'trust_connected_line' and 'send_connected_line' have been

added to the endpoint. The option 'trust_connected_line' is to control

if connected line updates are accepted from this endpoint.

The option 'send_connected_line' is to control if connected line updates

can be sent to this endpoint.

The default value is 'yes' for both options.

res_rtp_asterisk

------------------

* The existing strictrtp option in rtp.conf has a new choice availabe, called

'seqno', which behaves the same way as setting strictrtp to 'yes', but will

ignore the time interval during learning so that bursts of packets can still

trigger learning our source.

------------------------------------------------------------------------------

--- Functionality changes from Asterisk 15 to Asterisk 16 --------------------

------------------------------------------------------------------------------

app_fax

------------------

* The app_fax module is now deprecated, users should migrate to the

replacement module res_fax.

app_originate

------------------

* An 'a' option has been added to the Originate dialplan application which

will execute the originate in an asynchronous fashion. If set then the

application will return immediately without waiting for the originated

channel to answer.

Build System

------------------

* MALLOC_DEBUG no longer has an effect on Asterisk's ABI. Asterisk built

with MALLOC_DEBUG can now successfully load binary modules built without

MALLOC_DEBUG and vice versa. Third-party pre-compiled modules no longer

need to have a special build with it enabled.

* Asterisk now depends on libjansson >= 2.11. If this version is not

available on your distro you can use `./configure --with-jansson-bundled`.

app_macro

------------------

* The app_macro module is now deprecated and by default it is no longer

built. Users should migrate to app_stack (Gosub). A warning is logged

the first time any Macro is used.

app_setcallerid

------------------

* The app_setcallerid module has been removed. The CALLERID dialplan function

should be used instead.

chan_sip

------------------

* New function SIP_HEADERS() enumerates all headers in the incoming INVITE.

* The variable GET_TRANSFERRER_DATA set in the peer channel causes matching

headers be retrieved from the REFER message and made accessible to the

dialplan in the hash TRANSFER_DATA.

chan_dahdi

------------------

* Timeouts for reading digits from analog phones are now configurable in

chan_dahdi.conf: firstdigit_timeout, interdigit_timeout, matchdigit_timeout.

AMI

------------------

* The ContactStatus and Status fields for the manager events ContactStatus

and ContactStatusDetail are now set to "NonQualified" when a contact exists

but has not been qualified.

* The "Newexten" event is now part of the "dialplan" class. The documentation

for Asterisk 15 already specified this, but the implementation was actually

using the "call" class instead.

ARI

------------------

* The ContactInfo event's contact_status field is now set to "NonQualified"

when a contact exists but has not been qualified.

app_queue

------------------

* Added the ability to set the wrapuptime in the configuration of member.

When set the wrapuptime on the member is used instead of the wrapuptime

defined for the queue itself.

* Added predial handler support for caller and callee channels with the

B and b options respectively. This is similar to the predial support

in app_dial.

res_config_sqlite

------------------

* The res_config_sqlite module is now deprecated, users should migrate to the

replacement module res_config_sqlite3.

res_monitor

------------------

* The res_monitor module is now deprecated, users should migrate to the

replacement module app_mixmonitor.

res_pjsip

------------------

* A new AMI action, PJSIPShowAors, has been added which displays information

about all configured PJSIP AORs.

* A new AMI action, PJSIPShowAuths, has been added which displays information

about all configured PJSIP Auths.

* A new AMI action, PJSIPShowContacts, has been added which displays information

about all configured PJSIP Contacts.

res_pjsip_registrar_expire

------------------

* The res_pjsip_registrar_expire module has been removed. The functionality has

been moved into res_pjsip_registrar.

func_audiohookinherit

------------------

* The func_audiohookinherit module has been removed. Due to architectural changes

in Asterisk 12, audiohook inheritance is performed automatically and this

function now lacks function.

cdr_syslog

------------------

* The cdr_syslog module is now deprecated and by default it is no longer

built.

cdr_sqlite

------------------

* The cdr_sqlite module has been removed. Users should move to using the

cdr_sqlite3_custom module instead.

format_jpeg

------------------

* The format_jpeg module has been removed.

pbx_dundi

------------------

* DUNDi now supports IPv6

Core:

------------------

* libedit is no longer available as an embedded library and must be provided

by the system.

* The STATIC_BUILD functionality has been removed as it has not been maintained

and has not worked in quite some time.

* The module loader now enforces inter-module dependencies. This ensures that

a module is not started before another it depends on, even if preload is used.

If a dependency is not available or fails to startup this will block any

dependants from startup.

* Parts of the Asterisk core which can load configuration from realtime are now

built-in modules. It is no longer necessary to preload realtime drivers as

they are always initialized before the built-in modules.

------------------------------------------------------------------------------

--- Functionality changes from Asterisk 15.5.0 to Asterisk 15.6.0 ------------

------------------------------------------------------------------------------

res_pjsip

------------------

* A new option 'suppress_q850_reason_headers' has been added to the endpoint

object. Some devices can't accept multiple Reason headers and get confused

when both 'SIP' and 'Q.850' Reason headers are received. This option allows

the 'Q.850' Reason header to be suppressed. The default value is 'no'.

res_pjsip_endpoint_identifier_ip

------------------

* Added regex support to the identify section match_header option. You

specify a regex instead of an explicit string by surrounding the header

value with slashes:

match_header = SIPHeader: /regex/

------------------------------------------------------------------------------

--- Functionality changes from Asterisk 15.4.0 to Asterisk 15.5.0 ------------

------------------------------------------------------------------------------

Core

------------------

* Core bridging and, more specifically, bridge_softmix have been enhanced to

relay received frames of type TEXT or TEXT_DATA to all participants in a

softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to

take advantage of this so when res_pjsip_messaging receives an in-dialog

MESSAGE message from a user in a conference call, it's relayed to all

other participants in the call.

app_sendtext

------------------

* Support Enhanced Messaging. SendText now accepts new channel variables

that can be used to override the To and From display names and set the

Content-Type of a message. Since you can now set Content-Type, other

text/* content types are now valid.

app_confbridge

------------------

* ConfbridgeList now shows talking status. This utilizes the same voice

detection as the ConfbridgeTalking event, so bridges must be configured

with "talk_detection_events=yes" for this flag to have meaning.

* ConfBridge can now send events to participants via in-dialog MESSAGEs.

All current Confbridge events are supported, such as ConfbridgeJoin,

ConfbridgeLeave, etc. In addition to those events, a new event

ConfbridgeWelcome has been added that will send a list of all

current participants to a new participant.

res_pjsip

------------------

* Two new options have been added to the system and endpoint objects to

control whether, on outbound calls, Asterisk will accept updated SDP answers

during the initial INVITE transaction when 100rel is not in effect.

This usually happens when the INVITE is forked to multiple UASs and more

than one sends an SDP answer or when a single UAS needs to change a media

port to switch from custom ringback to the actual media destination.

The 'follow_early_media_forked' option sets whether Asterisk will accept

the updated SDP when the To tag on the subsequent response is different than

that on the the previous response. This usually occurs in the forked INVITE

scenario. The default value is "yes" which is the current behavior.

The 'accept_multiple_sdp_answers' flag sets whether Asterisk will accept the

updated SDP when the To tag on the subsequent response is the same as that

on the previous response. This can occur when a UAS needs to switch media

ports from custom ringback to the final media path. The default value is

"no" which is the current behavior.

These options have to be enabled system-wide in the system config section

of pjsip.conf as well as on individual endpoints that require the

functionality.

------------------------------------------------------------------------------

--- Functionality changes from Asterisk 15.3.0 to Asterisk 15.4.0 ------------

------------------------------------------------------------------------------

Core

------------------

* A new configuration option "genericplc_on_equal_codecs" was added to the

"plc" section of codecs.conf to allow generic packet loss concealment even

if no transcoding was originally needed. Transcoding via SLIN is forced

in this case.

res_pjproject

------------------

* 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.

res_pjsip_notify

------------------

* Extend the PJSIPNotify AMI command to send an in-dialog notify on a

channel.

------------------------------------------------------------------------------

--- Functionality changes from Asterisk 15.2.0 to Asterisk 15.3.0 ------------

------------------------------------------------------------------------------

Core

------------------

* During dialplan reload log messages are produced for each context,

extension and include. These messages are no longer printed by the

verbose loggers, they are now only logged as debug messages.

app_confbridge

------------------

* Added the Muted header to the ConfbridgeJoin AMI event to indicate the

participant's starting mute status.

* 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>".

app_followme

------------------

* Added a new prompt, connecting-prompt, which will be played

(if configured) to the "winner" callee before connecting the call.

res_pjsip

------------------

* Users who are matching endpoints by SIP header need to reevaluate their

global "endpoint_identifier_order" option in light of the "ip" endpoint

identifier method split into the "ip" and "header" endpoint identifier

methods.

* The pjsip_transport_event feature introduced in 15.1.0 has been refactored.

Any external modules that may have used that feature (highly unlikey) will

need to be changed as the API has been altered slightly.

res_pjsip_endpoint_identifier_ip

------------------

* The endpoint identifier "ip" method previously recognized endpoints either

by IP address or a matching SIP header. The "ip" endpoint identifier method

is now split into the "ip" and "header" endpoint identifier methods. The

"ip" endpoint identifier method only matches by IP address and the "header"

endpoint identifier method only matches by SIP header. The split allows the

user to control the relative priority of the IP address and the SIP header

identification methods in the global "endpoint_identifier_order" option.

e.g., If you have 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 can now predict which endpoint is matched when a request comes in that

matches both.

res_pjsip_pubsub

------------------

* 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. Since this

required the addition of a new column to the ps_subscription_persistence

realtime table, users who store their subscriptions in a database will

need to run the "alembic upgrade head" process to add the column to

the schema.

res_pjsip_transport_management

------------------

* 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.

------------------------------------------------------------------------------

--- Functionality changes from Asterisk 15.1.0 to Asterisk 15.2.0 ------------

------------------------------------------------------------------------------

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_rtp_asterisk

------------------

* The X.509 certificate used for DTLS negotation can now be automatically

generated. This is supported by res_pjsip by specifying

"dtls_auto_generate_cert = yes" on a PJSIP endpoint. For chan_sip, you

would set "dtlsautogeneratecert = yes" either in the [general] section of

sip.conf or on a specific peer.

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 15.0.0 to Asterisk 15.1.0 ------------

------------------------------------------------------------------------------

res_pjsip

------------------

* 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.

AMI

------------------

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

------------------------------------------------------------------------------

--- Functionality changes from Asterisk 14 to Asterisk 15 --------------------

------------------------------------------------------------------------------

app_queue

------------------

* PAUSEALL/UNPAUSEALL now sets the pause reason in the queue_log if it has

been defined.

* A new option, "announce-position-only-up," has been added that, when set to

yes, causes position announcements to only be played when the caller's

queue position has improved since the last time that we annouced their

position. This default is no.

Build System

------------------

* '--with-pjproject-bundled' is now the default when running ./configure

It can be disabled with '--without-pjproject-bundled'.

* 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.

------------------------------------------------------------------------------

--- Functionality changes from Asterisk 14.6.0 to Asterisk 14.7.0 ------------

------------------------------------------------------------------------------

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.

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.

* 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.

------------------------------------------------------------------------------

--- Functionality changes from Asterisk 14.5.0 to Asterisk 14.6.0 ------------

------------------------------------------------------------------------------

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 14.4.0 to Asterisk 14.5.0 ------------

------------------------------------------------------------------------------

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 14.3.0 to Asterisk 14.4.0 ------------

------------------------------------------------------------------------------

Build System

------------------

* LOW_MEMORY no longer has an effect on Asterisk ABI. Symbols that were

previously suppressed by LOW_MEMORY are now replaced by stub functions.

Asterisk built with LOW_MEMORY can now successfully load binary modules

built without LOW_MEMORY and vice versa.

* RADIUS backends for CEL and CDR can now also be built using the radcli

client library, in addition to the existing support for building them

using either freeradius or radiusclient-ng.

Core

------------------

* ASTERISK_REGISTER_FILE was no longer useful and has been removed. Sources

which use mtx_prof must now manually declare and initialize the variable.

chan_sip

------------------

* If an offer is received with optional SRTP (a media stream with RTP/AVP but

which contains a crypto line) chan_sip will now accept it and enable SRTP.

If you would like to do optional SRTP on outbound you will need to create

a dialplan that dials with it enabled initially and if it fails fall back to

without.

res_pjsip

------------------

* Added endpoint configuration parameter "preferred_codec_only".

This allow asterisk response to a SIP invite with the single most

preferred codec rather than advertising all joint codec capabilities.

This limits the other side's codec choice to exactly what we prefer.

cdr_radius

------------------

* To fix a memory leak the syslog channel is now empty if it has not been set

and used by a syslog channel in the logger.

cel_radius

------------------

* To fix a memory leak the syslog channel is now empty if it has not been set

and used by a syslog channel in the logger.

RTP

------------------

* New setting "rtp_pt_dynamic = 35" in asterisk.conf:

Normally the Dynamic RTP Payload Type numbers are 96-127, which allow just 32

formats. To avoid the message "No Dynamic RTP mapping available", the range

was changed to 35-63,96-127. This is allowed by RFC 3551 section 3. However,

when you use more than 32 formats and calls are not accepted by a remote

implementation, please report this and go back to rtp_pt_dynamic = 96.

* A new setting, "rtp_use_dynamic", has been added in asterisk.conf". When set

to "yes" RTP dynamic payload types are assigned dynamically per RTP instance.

When set to "no" RTP dynamic payload types are globally initialized to pre-

designated numbers and function similar to static payload types.

app_originate

------------------

* Added support to gosub predial routines on both original channel and on the

created channel using options parameter (like app_dial) B() and b(). This

allows for adding variables to newly created channel or, e.g. setting callerid.

CLI Commands

------------------

* 'dialplan show' output will now show [config_file:line_number] instead of

[registrar] when that information is available. Currently only extensions

registered by pbx_config when loading/reloading will use this format.

app_queue

------------------

* Add 'QueueUpdate' application which can be used to track outbound calls

using app_queue.

pbx_spool

------------------

* Asterisk will now set the AST_OUTGOING_ATTEMPT channel variable so that

attempt-specific behavior is possible. This is a 1-based number that

simply increases by 1 for each attempt.

------------------------------------------------------------------------------

--- Functionality changes from Asterisk 14.3.0 to Asterisk 14.4.0 ------------

------------------------------------------------------------------------------

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 14.2.0 to Asterisk 14.3.0 ------------

------------------------------------------------------------------------------

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 14.1.0 to Asterisk 14.2.0 ------------

------------------------------------------------------------------------------

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.

* The ARI major version has been bumped. There are not any known breaking changes

in ARI. The major version has been bumped because otherwise we can end up with

overlapping version numbers between different Asterisk versions. Now each major

version of Asterisk will bring with it a change in the major version of ARI.

The ARI version in Asterisk 14 is now 2.0.0.

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.

Queue

------------------

* A new dialplan variable, ABANDONED, is set when the call is not answered

by an agent.

res_ari

------------------

* The configuration file ari.conf now supports a channelvars option, which

specifies a list of channel variables to include in each channel-oriented

ARI event.

------------------------------------------------------------------------------

--- Functionality changes from Asterisk 14.0.0 to Asterisk 14.1.0 ------------

------------------------------------------------------------------------------

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"