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==============================================================================
Kevin P. Fleming
committed
===
=== 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.
===
==============================================================================
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--- Functionality changes from Asterisk 15 to Asterisk 16 --------------------
------------------------------------------------------------------------------
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.
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.
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.
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.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 15.1.0 to Asterisk 15.2.0 ------------
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Core
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* 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 ------------
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res_pjsip
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* 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 ------------
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res_pjsip
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* 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
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