|
|
|
|
@ -12,6 +12,112 @@
|
|
|
|
|
===
|
|
|
|
|
==============================================================================
|
|
|
|
|
|
|
|
|
|
------------------------------------------------------------------------------
|
|
|
|
|
--- Functionality changes from Asterisk 20.0.0 to Asterisk 20.1.0 ------------
|
|
|
|
|
------------------------------------------------------------------------------
|
|
|
|
|
|
|
|
|
|
AMI
|
|
|
|
|
------------------
|
|
|
|
|
* The AOCMessage action can now be used to generate AOC-S messages.
|
|
|
|
|
|
|
|
|
|
Add support for named capture agent.
|
|
|
|
|
------------------
|
|
|
|
|
* A name for the capture agent can now be specified
|
|
|
|
|
using the capture_name option which, if specified,
|
|
|
|
|
will be sent to the HEP server.
|
|
|
|
|
|
|
|
|
|
app_if
|
|
|
|
|
------------------
|
|
|
|
|
* Adds the If, ElseIf, Else, EndIf, and ExitIf applications
|
|
|
|
|
for conditional execution of a block of code.
|
|
|
|
|
|
|
|
|
|
app_mixmonitor
|
|
|
|
|
------------------
|
|
|
|
|
* The d option for MixMonitor now allows deleting
|
|
|
|
|
the original recording when MixMonitor exits,
|
|
|
|
|
which can be useful when MixMonitor copies it
|
|
|
|
|
somewhere else before exiting.
|
|
|
|
|
|
|
|
|
|
* Adds the c option to use the real Caller ID on
|
|
|
|
|
the channel in voicemail recordings as opposed
|
|
|
|
|
to the Connected Line.
|
|
|
|
|
|
|
|
|
|
app_voicemail
|
|
|
|
|
------------------
|
|
|
|
|
* The voicemail user option attachextrecs can
|
|
|
|
|
now be set to control whether external recordings
|
|
|
|
|
trigger voicemail email notifications.
|
|
|
|
|
|
|
|
|
|
cdr
|
|
|
|
|
------------------
|
|
|
|
|
* Two new options have been added which allow
|
|
|
|
|
bridging and dial state changes to be ignored
|
|
|
|
|
in CDRs, which can be useful if a single CDR
|
|
|
|
|
is desired for a channel.
|
|
|
|
|
|
|
|
|
|
chan_dahdi
|
|
|
|
|
------------------
|
|
|
|
|
* FXO channels (FXS signaled) that don't use callerid or
|
|
|
|
|
distinctive ring detection can now be configured
|
|
|
|
|
to enter the dialplan immediately using immediate=yes,
|
|
|
|
|
instead of waiting for at least one ring.
|
|
|
|
|
|
|
|
|
|
pbx_builtins
|
|
|
|
|
------------------
|
|
|
|
|
* It is now possible to not wait for media on
|
|
|
|
|
a channel when answering it using Answer,
|
|
|
|
|
by specifying the i option.
|
|
|
|
|
|
|
|
|
|
res_pjsip
|
|
|
|
|
------------------
|
|
|
|
|
* Added options "security_negotiation" and "security_mechanisms" to pjsip
|
|
|
|
|
endpoints and registrations. "security_negotiation" can be set to "no" (default)
|
|
|
|
|
or "mediasec", and "security_mechanisms" can be a list of comma-separated
|
|
|
|
|
security_mechanisms in the form defined by RFC 3329 section 2.2.
|
|
|
|
|
|
|
|
|
|
* A new option named "all_codecs_on_empty_reinvite" has been added to the
|
|
|
|
|
global section. When this option is enabled, on reception of a re-INVITE
|
|
|
|
|
without SDP, Asterisk will send an SDP offer in the 200 OK response containing
|
|
|
|
|
all configured codecs on the endpoint, instead of simply those that have
|
|
|
|
|
already been negotiated. RFC 3261 specifies this as a SHOULD requirement.
|
|
|
|
|
The default value is "off".
|
|
|
|
|
|
|
|
|
|
res_pjsip_aoc
|
|
|
|
|
------------------
|
|
|
|
|
* Added res_pjsip_aoc which gives chan_pjsip the ability to send Advice-of-Charge messages.
|
|
|
|
|
A new endpoint option, send_aoc, controls this.
|
|
|
|
|
|
|
|
|
|
res_pjsip_header_funcs
|
|
|
|
|
------------------
|
|
|
|
|
* The new PJSIP_HEADER_PARAM function now fully supports both
|
|
|
|
|
URI and header parameters. Both reading and writing
|
|
|
|
|
parameters are supported.
|
|
|
|
|
|
|
|
|
|
res_pjsip_logger
|
|
|
|
|
------------------
|
|
|
|
|
* SIP messages can now be filtered by SIP request method
|
|
|
|
|
(INVITE, CANCEL, ACK, BYE, REGISTER, OPTION,
|
|
|
|
|
SUBSCRIBE, NOTIFY, PUBLISH, INFO, and MESSAGE),
|
|
|
|
|
allowing for more granular debugging to be done
|
|
|
|
|
in the CLI. This applies to requests but not responses.
|
|
|
|
|
|
|
|
|
|
res_pjsip_notify
|
|
|
|
|
------------------
|
|
|
|
|
* Allows using the config options in pjsip_notify.conf
|
|
|
|
|
from AMI actions as with the existing CLI commands.
|
|
|
|
|
|
|
|
|
|
res_tonedetect
|
|
|
|
|
------------------
|
|
|
|
|
* The TONE_DETECT function now supports
|
|
|
|
|
detection of audible ringback tone
|
|
|
|
|
using the p option.
|
|
|
|
|
|
|
|
|
|
xmldocs
|
|
|
|
|
------------------
|
|
|
|
|
* The XML documentation can now be reloaded without restarting
|
|
|
|
|
Asterisk, which makes it possible to load new modules that
|
|
|
|
|
enforce documentation without restarting Asterisk.
|
|
|
|
|
|
|
|
|
|
------------------------------------------------------------------------------
|
|
|
|
|
--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------
|
|
|
|
|
------------------------------------------------------------------------------
|
|
|
|
|
|
Asterisk Development Team
2 years ago