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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.
===
==============================================================================
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 16.0.0 to Asterisk 16.1.0 ------------
------------------------------------------------------------------------------
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
* 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
------------------
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