mirror of
https://github.com/asterisk/asterisk.git
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Update CHANGES and UPGRADE.txt for 19.0.0
This commit is contained in:
Asterisk Development Team
272
CHANGES
272
CHANGES
@@ -12,6 +12,278 @@
|
||||
===
|
||||
==============================================================================
|
||||
|
||||
------------------------------------------------------------------------------
|
||||
--- Functionality changes from Asterisk 18.0.0 to Asterisk 19.0.0 ------------
|
||||
------------------------------------------------------------------------------
|
||||
|
||||
AMI Flash event
|
||||
------------------
|
||||
* Hook flash events are now exposed as AMI events.
|
||||
|
||||
Add variable support to Originate
|
||||
------------------
|
||||
* The Originate application now allows
|
||||
variables to be set on the new channel
|
||||
through a new option.
|
||||
|
||||
Core
|
||||
------------------
|
||||
* Added debug logging categories that allow a user to output debug information
|
||||
based on a specified category. This lets the user limit, and filter debug
|
||||
output to data relevant to a particular context, or topic. For instance the
|
||||
following categories are now available for debug logging purposes:
|
||||
|
||||
dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet
|
||||
|
||||
These debug categories can be enable/disable via an Asterisk CLI command:
|
||||
|
||||
core set debug category <category>[:<sublevel>] [category[:<sublevel] ...]
|
||||
core set debug category off [<category> [<category>] ...]
|
||||
|
||||
If no sub-level is associated all debug statements for a given category are
|
||||
output. If a sub-level is given then only those statements assigned a value
|
||||
at or below the associated sub-level are output.
|
||||
|
||||
* The location where the media cache stores its temporary files
|
||||
is no longer hardcoded to /tmp but can now be configured separately
|
||||
via the astcachedir config variable in asterisk.conf.
|
||||
|
||||
The default location for astcachedir is now /var/cache/asterisk
|
||||
instead of /tmp, please make sure to manually cleanup and/or
|
||||
migrate the temporary files in /tmp after upgrading.
|
||||
|
||||
MessageSend
|
||||
------------------
|
||||
* The MessageSend dialplan application now takes an
|
||||
optional third argument that can set the message's
|
||||
"To" field on outgoing messages. It's an alternative
|
||||
to using the MESSAGE(to) dialplan function.
|
||||
|
||||
To prevent confusion with the first argument, currently
|
||||
named "to", it's been renamed to "destination".
|
||||
Its function, creating the request URI, hasn't changed.
|
||||
|
||||
The online documentation has also been enhanced to
|
||||
explain the behavior.
|
||||
|
||||
Despite the changes in this commit, there should be
|
||||
no impact to current users of MessageSend.
|
||||
|
||||
New ConfKick application
|
||||
------------------
|
||||
* Adds a ConfKick() application, which allows
|
||||
a specific channel, all users, or all non-admin
|
||||
users to be kicked from a conference bridge.
|
||||
|
||||
New Reload application
|
||||
------------------
|
||||
* Adds an application to reload modules
|
||||
|
||||
PlaybackFinished has a new error state
|
||||
------------------
|
||||
* The PlaybackFinished event now has a new state "failed"
|
||||
that is used when the sound file was not played due to an error.
|
||||
Before the state on PlaybackFinished was always "done".
|
||||
|
||||
In case of multiple sound files to be played,
|
||||
the PlaybackFinished is sent only once in the end of the list,
|
||||
even in case of error.
|
||||
|
||||
WaitForCondition application
|
||||
------------------
|
||||
* This application provides a way to halt
|
||||
dialplan execution until a provided
|
||||
condition evaluates to true.
|
||||
|
||||
app_confbridge
|
||||
------------------
|
||||
* app_confbridge now has the ability to force the estimated bitrate on an SFU
|
||||
bridge. To use it, set a bridge profile's remb_behavior to "force" and
|
||||
set remb_estimated_bitrate to a rate in bits per second. The
|
||||
remb_estimated_bitrate parameter is ignored if remb_behavior is something
|
||||
other than "force".
|
||||
|
||||
app_confbridge answer supervision control
|
||||
------------------
|
||||
* app_confbridge now provides a user option to prevent
|
||||
answer supervision if the channel hasn't been
|
||||
answered yet. To use it, set a user profile's
|
||||
answer_channel option to no.
|
||||
|
||||
app_dial announcement option
|
||||
------------------
|
||||
* The A option for Dial now supports
|
||||
playing audio to the caller as well
|
||||
as the called party.
|
||||
|
||||
app_mixmonitor
|
||||
------------------
|
||||
* app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and
|
||||
MixMonitorMute when the channel monitoring is started, stopped and muted (or
|
||||
unmuted) respectively.
|
||||
|
||||
app_voicemail
|
||||
------------------
|
||||
* The VoiceMail application can now be configured to send greetings and
|
||||
instructions via early media and only answering the channel when it is
|
||||
time for the caller to record their message. This behavior can be
|
||||
activated by passing the new 'e' option to VoiceMail.
|
||||
|
||||
* You can now customize the "beep" tone or omit it entirely.
|
||||
|
||||
chan_iax2
|
||||
------------------
|
||||
* You can now specify a default "auth" method in the
|
||||
[general] section of iax.conf
|
||||
|
||||
chan_pjsip
|
||||
------------------
|
||||
* The PJSIP_SEND_SESSION_REFRESH dialplan function now issues a warning, and
|
||||
returns unsuccessful if it's used on a channel prior to answering.
|
||||
|
||||
chan_pjsip, app_transfer
|
||||
------------------
|
||||
* Added TRANSFERSTATUSPROTOCOL variable. When transfer is performed,
|
||||
transfers can pass a protocol specific error code.
|
||||
Example, in SIP 3xx-6xx represent any SIP specific error received when
|
||||
performing a REFER.
|
||||
|
||||
func_math: Three new dialplan functions
|
||||
------------------
|
||||
* Introduce three new functions, MIN, MAX, and ABS, which can be used to
|
||||
obtain the minimum or maximum of up to two integers or absolute value.
|
||||
|
||||
func_odbc
|
||||
------------------
|
||||
* Introduce an ARGC variable for func_odbc functions, along with a minargs
|
||||
per-function configuration option.
|
||||
|
||||
minargs enables enforcing of minimum count of arguments to pass to
|
||||
func_odbc, so if you're unconditionally using ARG1 through ARG4 then
|
||||
this should be set to 4. func_odbc will generate an error in this case,
|
||||
so for example
|
||||
|
||||
[FOO]
|
||||
minargs = 4
|
||||
|
||||
and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
|
||||
potentially leaked ARG4 from Gosub().
|
||||
|
||||
ARGC is needed if you're using optional argument, to verify whether or
|
||||
not an argument has been passed, else it's possible to use a leaked ARGn
|
||||
from Gosub (app_stack). So now you can safely do
|
||||
${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.
|
||||
|
||||
func_volume now can be read
|
||||
------------------
|
||||
* The VOLUME function can now also be used
|
||||
to read existing values previously set.
|
||||
|
||||
logger
|
||||
------------------
|
||||
* Added a new log formatter called "plain" that always prints
|
||||
file, function and line number if available (even for verbose
|
||||
messages) and never prints color control characters. Most
|
||||
suitable for file output but can be used for other channels
|
||||
as well.
|
||||
|
||||
You use it in logger.conf like so:
|
||||
debug => [plain]debug
|
||||
console => [plain]error,warning,debug,notice,pjsip_history
|
||||
messages => [plain]warning,error,verbose
|
||||
|
||||
* The dateformat option in logger.conf will now control the remote
|
||||
console (asterisk -r -T) timestamp format. Previously, dateformat only
|
||||
controlled the formatting of the timestamp going to log files and the
|
||||
main console (asterisk -c) but only for non-verbose messages.
|
||||
|
||||
Internally, Asterisk does not send the logging timestamp with verbose
|
||||
messages to console clients. It's up to the Asterisk remote consoles
|
||||
to format verbose messages. Asterisk remote consoles previously did
|
||||
not load dateformat from logger.conf.
|
||||
|
||||
Previously there was a non-configurable and hard-coded "%b %e %T"
|
||||
dateformat that would be used no matter what on all verbose console
|
||||
messages printed on remote consoles.
|
||||
|
||||
Example:
|
||||
logger.conf
|
||||
dateformat=%F %T.%3q
|
||||
|
||||
# asterisk -rvvv -T
|
||||
[2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so.
|
||||
[Mar 19 09:55:43] -- Goto (dialExten,s,1)
|
||||
|
||||
Given the following example configuration in logger.conf, Asterisk log
|
||||
files and the console, will log verbose messages using the given
|
||||
timestamp. Now ensuring that all remote console messages are logged
|
||||
with the same dateformat as other log streams.
|
||||
|
||||
---
|
||||
[general]
|
||||
dateformat=%F %T.%3q
|
||||
|
||||
[logfiles]
|
||||
console => notice,warning,error,verbose
|
||||
full => notice,warning,error,debug,verbose
|
||||
---
|
||||
|
||||
Now we have a globally-defined dateformat that will be used
|
||||
consistently across the Asterisk main console, remote consoles, and
|
||||
log files.
|
||||
|
||||
Now we have consistent logging:
|
||||
|
||||
# asterisk -rvvv -T
|
||||
[2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so.
|
||||
[2021-03-19 09:55:43.920-0400] -- Goto (dialExten,s,1)
|
||||
|
||||
res_pjsip
|
||||
------------------
|
||||
* PJSIP transports can now be partially reloaded safely. This allows the
|
||||
local_net and external_* options to be updated without restarting Asterisk.
|
||||
|
||||
* PJSIP endpoints can now be configured to skip authentication when
|
||||
handling OPTIONS requests by setting the allow_unauthenticated_options
|
||||
configuration property to 'yes.'
|
||||
|
||||
* PJSIP support of registrations of endpoints in multidomain
|
||||
scenarios, where the endpoint contains the domain info
|
||||
in pjsip_endpoint.conf
|
||||
|
||||
res_pjsip_dialog_info_body_generator
|
||||
------------------
|
||||
* PJSIP now supports RFC 4235 Section 4.1.6 dialog-info+xml local and
|
||||
remote elements by iterating through ringing channels and inserting
|
||||
that info into NOTIFY packet sent to the endpoint.
|
||||
|
||||
res_pjsip_messaging
|
||||
------------------
|
||||
* Implemented the new "to" parameter of the MessageSend()
|
||||
dialplan application. This allows a user to specify
|
||||
a complete SIP "To" header separate from the Request URI.
|
||||
We now also accept a destination in the same format
|
||||
as Dial()... PJSIP/number@endpoint
|
||||
|
||||
res_rtp_asterisk
|
||||
------------------
|
||||
* By default Asterisk reports the PJSIP version in all
|
||||
STUN packets it sends.
|
||||
|
||||
This behaviour may not be desired in a production
|
||||
environment and can now be disabled by setting the
|
||||
stun_software_attribute option to 'no' in rtp.conf.
|
||||
|
||||
res_srtp
|
||||
------------------
|
||||
* SRTP replay protection has been added to res_srtp and
|
||||
a new configuration option "srtpreplayprotection" has
|
||||
been added to the rtp.conf config file. For security
|
||||
reasons, the default setting is "yes". Buggy clients
|
||||
may not handle this correctly which could result in
|
||||
no, or one way, audio and Asterisk error messages like
|
||||
"replay check failed".
|
||||
|
||||
------------------------------------------------------------------------------
|
||||
--- New functionality introduced in Asterisk 18.0.0 --------------------------
|
||||
------------------------------------------------------------------------------
|
||||
|
||||
52
UPGRADE.txt
52
UPGRADE.txt
@@ -18,6 +18,58 @@
|
||||
===
|
||||
===========================================================
|
||||
|
||||
------------------------------------------------------------------------------
|
||||
--- New functionality introduced in Asterisk 19.0.0 --------------------------
|
||||
------------------------------------------------------------------------------
|
||||
|
||||
Log Rotate
|
||||
------------------
|
||||
* The sample logger files have been changed to have .log as their file
|
||||
extension. This was done so that when attached to issues on the issue
|
||||
tracker, they are able to be opened in the browser for convenience.
|
||||
Because of this, the asterisk.logrotate script has been updated to look
|
||||
for .log extensions instead of no extension for files such as full
|
||||
and messages.
|
||||
|
||||
chan_sip
|
||||
------------------
|
||||
* chan_sip is no longer built by default. To build it, make sure to
|
||||
enable it when running 'make menuselect'
|
||||
|
||||
------------------------------------------------------------------------------
|
||||
--- Functionality changes from Asterisk 18.0.0 to Asterisk 19.0.0 ------------
|
||||
------------------------------------------------------------------------------
|
||||
|
||||
STIR/SHAKEN
|
||||
------------------
|
||||
* The configuration option public_key_url in stir_shaken.conf
|
||||
has been renamed to public_cert_url to better fit what it
|
||||
contains. Only the name has changed - functionality is the
|
||||
same.
|
||||
|
||||
* STIR/SHAKEN originally needed an origid to be specified in
|
||||
stir_shaken.conf under the certificate config object in
|
||||
order to work. Now, one is automatically created by
|
||||
generating a UUID, as recommended by RFC8588. Any origid
|
||||
you have in your stir_shaken.conf will need to be removed
|
||||
for the module to read in certificates.
|
||||
|
||||
menuselect
|
||||
------------------
|
||||
* menuselect --enable, --disable, --enable-category and --disable-category will
|
||||
now fail with a non-zero exit code instead of silently failing if an invalid
|
||||
option or category is specified.
|
||||
|
||||
res_srtp
|
||||
------------------
|
||||
* SRTP replay protection has been added to res_srtp and
|
||||
a new configuration option "srtpreplayprotection" has
|
||||
been added to the rtp.conf config file. For security
|
||||
reasons, the default setting is "yes". Buggy clients
|
||||
may not handle this correctly which could result in
|
||||
no, or one way, audio and Asterisk error messages like
|
||||
"replay check failed".
|
||||
|
||||
------------------------------------------------------------------------------
|
||||
--- New functionality introduced in Asterisk 18.0.0 --------------------------
|
||||
------------------------------------------------------------------------------
|
||||
|
||||
@@ -1,7 +0,0 @@
|
||||
Subject: app_confbridge
|
||||
|
||||
app_confbridge now has the ability to force the estimated bitrate on an SFU
|
||||
bridge. To use it, set a bridge profile's remb_behavior to "force" and
|
||||
set remb_estimated_bitrate to a rate in bits per second. The
|
||||
remb_estimated_bitrate parameter is ignored if remb_behavior is something
|
||||
other than "force".
|
||||
@@ -1,6 +0,0 @@
|
||||
Subject: app_confbridge answer supervision control
|
||||
|
||||
app_confbridge now provides a user option to prevent
|
||||
answer supervision if the channel hasn't been
|
||||
answered yet. To use it, set a user profile's
|
||||
answer_channel option to no.
|
||||
@@ -1,6 +0,0 @@
|
||||
Subject: New ConfKick application
|
||||
|
||||
Adds a ConfKick() application, which allows
|
||||
a specific channel, all users, or all non-admin
|
||||
users to be kicked from a conference bridge.
|
||||
|
||||
@@ -1,6 +0,0 @@
|
||||
Subject: app_dial announcement option
|
||||
|
||||
The A option for Dial now supports
|
||||
playing audio to the caller as well
|
||||
as the called party.
|
||||
|
||||
@@ -1,6 +0,0 @@
|
||||
Subject: Add variable support to Originate
|
||||
|
||||
The Originate application now allows
|
||||
variables to be set on the new channel
|
||||
through a new option.
|
||||
|
||||
@@ -1,4 +0,0 @@
|
||||
Subject: New Reload application
|
||||
|
||||
Adds an application to reload modules
|
||||
|
||||
@@ -1,6 +0,0 @@
|
||||
Subject: chan_pjsip, app_transfer
|
||||
|
||||
Added TRANSFERSTATUSPROTOCOL variable. When transfer is performed,
|
||||
transfers can pass a protocol specific error code.
|
||||
Example, in SIP 3xx-6xx represent any SIP specific error received when
|
||||
performing a REFER.
|
||||
@@ -1,5 +0,0 @@
|
||||
Subject: WaitForCondition application
|
||||
|
||||
This application provides a way to halt
|
||||
dialplan execution until a provided
|
||||
condition evaluates to true.
|
||||
@@ -1,4 +0,0 @@
|
||||
Subject: chan_iax2
|
||||
|
||||
You can now specify a default "auth" method in the
|
||||
[general] section of iax.conf
|
||||
@@ -1,3 +0,0 @@
|
||||
Subject: AMI Flash event
|
||||
|
||||
Hook flash events are now exposed as AMI events.
|
||||
@@ -1,4 +0,0 @@
|
||||
Subject: func_math: Three new dialplan functions
|
||||
|
||||
Introduce three new functions, MIN, MAX, and ABS, which can be used to
|
||||
obtain the minimum or maximum of up to two integers or absolute value.
|
||||
@@ -1,20 +0,0 @@
|
||||
Subject: func_odbc
|
||||
|
||||
Introduce an ARGC variable for func_odbc functions, along with a minargs
|
||||
per-function configuration option.
|
||||
|
||||
minargs enables enforcing of minimum count of arguments to pass to
|
||||
func_odbc, so if you're unconditionally using ARG1 through ARG4 then
|
||||
this should be set to 4. func_odbc will generate an error in this case,
|
||||
so for example
|
||||
|
||||
[FOO]
|
||||
minargs = 4
|
||||
|
||||
and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
|
||||
potentially leaked ARG4 from Gosub().
|
||||
|
||||
ARGC is needed if you're using optional argument, to verify whether or
|
||||
not an argument has been passed, else it's possible to use a leaked ARGn
|
||||
from Gosub (app_stack). So now you can safely do
|
||||
${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.
|
||||
@@ -1,4 +0,0 @@
|
||||
Subject: func_volume now can be read
|
||||
|
||||
The VOLUME function can now also be used
|
||||
to read existing values previously set.
|
||||
@@ -1,18 +0,0 @@
|
||||
Subject: Core
|
||||
|
||||
Added debug logging categories that allow a user to output debug information
|
||||
based on a specified category. This lets the user limit, and filter debug
|
||||
output to data relevant to a particular context, or topic. For instance the
|
||||
following categories are now available for debug logging purposes:
|
||||
|
||||
dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet
|
||||
|
||||
These debug categories can be enable/disable via an Asterisk CLI command:
|
||||
|
||||
core set debug category <category>[:<sublevel>] [category[:<sublevel] ...]
|
||||
core set debug category off [<category> [<category>] ...]
|
||||
|
||||
If no sub-level is associated all debug statements for a given category are
|
||||
output. If a sub-level is given then only those statements assigned a value
|
||||
at or below the associated sub-level are output.
|
||||
|
||||
@@ -1,47 +0,0 @@
|
||||
Subject: logger
|
||||
|
||||
The dateformat option in logger.conf will now control the remote
|
||||
console (asterisk -r -T) timestamp format. Previously, dateformat only
|
||||
controlled the formatting of the timestamp going to log files and the
|
||||
main console (asterisk -c) but only for non-verbose messages.
|
||||
|
||||
Internally, Asterisk does not send the logging timestamp with verbose
|
||||
messages to console clients. It's up to the Asterisk remote consoles
|
||||
to format verbose messages. Asterisk remote consoles previously did
|
||||
not load dateformat from logger.conf.
|
||||
|
||||
Previously there was a non-configurable and hard-coded "%b %e %T"
|
||||
dateformat that would be used no matter what on all verbose console
|
||||
messages printed on remote consoles.
|
||||
|
||||
Example:
|
||||
logger.conf
|
||||
dateformat=%F %T.%3q
|
||||
|
||||
# asterisk -rvvv -T
|
||||
[2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so.
|
||||
[Mar 19 09:55:43] -- Goto (dialExten,s,1)
|
||||
|
||||
Given the following example configuration in logger.conf, Asterisk log
|
||||
files and the console, will log verbose messages using the given
|
||||
timestamp. Now ensuring that all remote console messages are logged
|
||||
with the same dateformat as other log streams.
|
||||
|
||||
---
|
||||
[general]
|
||||
dateformat=%F %T.%3q
|
||||
|
||||
[logfiles]
|
||||
console => notice,warning,error,verbose
|
||||
full => notice,warning,error,debug,verbose
|
||||
---
|
||||
|
||||
Now we have a globally-defined dateformat that will be used
|
||||
consistently across the Asterisk main console, remote consoles, and
|
||||
log files.
|
||||
|
||||
Now we have consistent logging:
|
||||
|
||||
# asterisk -rvvv -T
|
||||
[2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so.
|
||||
[2021-03-19 09:55:43.920-0400] -- Goto (dialExten,s,1)
|
||||
@@ -1,12 +0,0 @@
|
||||
Subject: logger
|
||||
|
||||
Added a new log formatter called "plain" that always prints
|
||||
file, function and line number if available (even for verbose
|
||||
messages) and never prints color control characters. Most
|
||||
suitable for file output but can be used for other channels
|
||||
as well.
|
||||
|
||||
You use it in logger.conf like so:
|
||||
debug => [plain]debug
|
||||
console => [plain]error,warning,debug,notice,pjsip_history
|
||||
messages => [plain]warning,error,verbose
|
||||
@@ -1,9 +0,0 @@
|
||||
Subject: Core
|
||||
|
||||
The location where the media cache stores its temporary files
|
||||
is no longer hardcoded to /tmp but can now be configured separately
|
||||
via the astcachedir config variable in asterisk.conf.
|
||||
|
||||
The default location for astcachedir is now /var/cache/asterisk
|
||||
instead of /tmp, please make sure to manually cleanup and/or
|
||||
migrate the temporary files in /tmp after upgrading.
|
||||
@@ -1,16 +0,0 @@
|
||||
Subject: MessageSend
|
||||
|
||||
The MessageSend dialplan application now takes an
|
||||
optional third argument that can set the message's
|
||||
"To" field on outgoing messages. It's an alternative
|
||||
to using the MESSAGE(to) dialplan function.
|
||||
|
||||
To prevent confusion with the first argument, currently
|
||||
named "to", it's been renamed to "destination".
|
||||
Its function, creating the request URI, hasn't changed.
|
||||
|
||||
The online documentation has also been enhanced to
|
||||
explain the behavior.
|
||||
|
||||
Despite the changes in this commit, there should be
|
||||
no impact to current users of MessageSend.
|
||||
@@ -1,5 +0,0 @@
|
||||
Subject: app_mixmonitor
|
||||
|
||||
app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and
|
||||
MixMonitorMute when the channel monitoring is started, stopped and muted (or
|
||||
unmuted) respectively.
|
||||
@@ -1,5 +0,0 @@
|
||||
Subject: res_pjsip
|
||||
|
||||
PJSIP endpoints can now be configured to skip authentication when
|
||||
handling OPTIONS requests by setting the allow_unauthenticated_options
|
||||
configuration property to 'yes.'
|
||||
@@ -1,4 +0,0 @@
|
||||
Subject: chan_pjsip
|
||||
|
||||
The PJSIP_SEND_SESSION_REFRESH dialplan function now issues a warning, and
|
||||
returns unsuccessful if it's used on a channel prior to answering.
|
||||
@@ -1,4 +0,0 @@
|
||||
Subject: res_pjsip
|
||||
|
||||
PJSIP transports can now be partially reloaded safely. This allows the
|
||||
local_net and external_* options to be updated without restarting Asterisk.
|
||||
@@ -1,5 +0,0 @@
|
||||
Subject: res_pjsip
|
||||
|
||||
PJSIP support of registrations of endpoints in multidomain
|
||||
scenarios, where the endpoint contains the domain info
|
||||
in pjsip_endpoint.conf
|
||||
@@ -1,5 +0,0 @@
|
||||
Subject: res_pjsip_dialog_info_body_generator
|
||||
|
||||
PJSIP now supports RFC 4235 Section 4.1.6 dialog-info+xml local and
|
||||
remote elements by iterating through ringing channels and inserting
|
||||
that info into NOTIFY packet sent to the endpoint.
|
||||
@@ -1,5 +0,0 @@
|
||||
res_pjsip_dtmf_info: Hook flash
|
||||
|
||||
Adds recognition for application/
|
||||
hook-flash as a hook flash event.
|
||||
|
||||
@@ -1,7 +0,0 @@
|
||||
Subject: res_pjsip_messaging
|
||||
|
||||
Implemented the new "to" parameter of the MessageSend()
|
||||
dialplan application. This allows a user to specify
|
||||
a complete SIP "To" header separate from the Request URI.
|
||||
We now also accept a destination in the same format
|
||||
as Dial()... PJSIP/number@endpoint
|
||||
@@ -1,8 +0,0 @@
|
||||
Subject: res_rtp_asterisk
|
||||
|
||||
By default Asterisk reports the PJSIP version in all
|
||||
STUN packets it sends.
|
||||
|
||||
This behaviour may not be desired in a production
|
||||
environment and can now be disabled by setting the
|
||||
stun_software_attribute option to 'no' in rtp.conf.
|
||||
@@ -1,9 +0,0 @@
|
||||
Subject: PlaybackFinished has a new error state
|
||||
|
||||
The PlaybackFinished event now has a new state "failed"
|
||||
that is used when the sound file was not played due to an error.
|
||||
Before the state on PlaybackFinished was always "done".
|
||||
|
||||
In case of multiple sound files to be played,
|
||||
the PlaybackFinished is sent only once in the end of the list,
|
||||
even in case of error.
|
||||
@@ -1,9 +0,0 @@
|
||||
Subject: res_srtp
|
||||
|
||||
SRTP replay protection has been added to res_srtp and
|
||||
a new configuration option "srtpreplayprotection" has
|
||||
been added to the rtp.conf config file. For security
|
||||
reasons, the default setting is "yes". Buggy clients
|
||||
may not handle this correctly which could result in
|
||||
no, or one way, audio and Asterisk error messages like
|
||||
"replay check failed".
|
||||
@@ -1,3 +0,0 @@
|
||||
Subject: app_voicemail
|
||||
|
||||
You can now customize the "beep" tone or omit it entirely.
|
||||
@@ -1,6 +0,0 @@
|
||||
Subject: app_voicemail
|
||||
|
||||
The VoiceMail application can now be configured to send greetings and
|
||||
instructions via early media and only answering the channel when it is
|
||||
time for the caller to record their message. This behavior can be
|
||||
activated by passing the new 'e' option to VoiceMail.
|
||||
@@ -1,9 +0,0 @@
|
||||
Subject: Log Rotate
|
||||
Master-Only: True
|
||||
|
||||
The sample logger files have been changed to have .log as their file
|
||||
extension. This was done so that when attached to issues on the issue
|
||||
tracker, they are able to be opened in the browser for convenience.
|
||||
Because of this, the asterisk.logrotate script has been updated to look
|
||||
for .log extensions instead of no extension for files such as full
|
||||
and messages.
|
||||
@@ -1,5 +0,0 @@
|
||||
Subject: chan_sip
|
||||
Master-Only: True
|
||||
|
||||
chan_sip is no longer built by default. To build it, make sure to
|
||||
enable it when running 'make menuselect'
|
||||
@@ -1,5 +0,0 @@
|
||||
Subject: menuselect
|
||||
|
||||
menuselect --enable, --disable, --enable-category and --disable-category will
|
||||
now fail with a non-zero exit code instead of silently failing if an invalid
|
||||
option or category is specified.
|
||||
@@ -1,9 +0,0 @@
|
||||
Subject: res_srtp
|
||||
|
||||
SRTP replay protection has been added to res_srtp and
|
||||
a new configuration option "srtpreplayprotection" has
|
||||
been added to the rtp.conf config file. For security
|
||||
reasons, the default setting is "yes". Buggy clients
|
||||
may not handle this correctly which could result in
|
||||
no, or one way, audio and Asterisk error messages like
|
||||
"replay check failed".
|
||||
@@ -1,6 +0,0 @@
|
||||
Subject: STIR/SHAKEN
|
||||
|
||||
The configuration option public_key_url in stir_shaken.conf
|
||||
has been renamed to public_cert_url to better fit what it
|
||||
contains. Only the name has changed - functionality is the
|
||||
same.
|
||||
@@ -1,8 +0,0 @@
|
||||
Subject: STIR/SHAKEN
|
||||
|
||||
STIR/SHAKEN originally needed an origid to be specified in
|
||||
stir_shaken.conf under the certificate config object in
|
||||
order to work. Now, one is automatically created by
|
||||
generating a UUID, as recommended by RFC8588. Any origid
|
||||
you have in your stir_shaken.conf will need to be removed
|
||||
for the module to read in certificates.
|
||||
Reference in New Issue
Block a user