Newer
Older
==============================================================================
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.
===
==============================================================================
Matthew Jordan
committed
------------------------------------------------------------------------------
--- 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 ----------
------------------------------------------------------------------------------
Loading
Loading full blame...