Update CHANGES and UPGRADE.txt for 18.3.0
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CHANGES
53
CHANGES
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@ -12,6 +12,59 @@
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===
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==============================================================================
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 18.2.2 to Asterisk 18.3.0 ------------
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------------------------------------------------------------------------------
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app_mixmonitor
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------------------
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* app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and
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MixMonitorMute when the channel monitoring is started, stopped and muted (or
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unmuted) respectively.
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chan_iax2
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------------------
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* You can now specify a default "auth" method in the
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[general] section of iax.conf
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chan_pjsip, app_transfer
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------------------
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* Added TRANSFERSTATUSPROTOCOL variable. When transfer is performed,
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transfers can pass a protocol specific error code.
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Example, in SIP 3xx-6xx represent any SIP specific error received when
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performing a REFER.
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func_odbc
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------------------
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* Introduce an ARGC variable for func_odbc functions, along with a minargs
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per-function configuration option.
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minargs enables enforcing of minimum count of arguments to pass to
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func_odbc, so if you're unconditionally using ARG1 through ARG4 then
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this should be set to 4. func_odbc will generate an error in this case,
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so for example
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[FOO]
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minargs = 4
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and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
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potentially leaked ARG4 from Gosub().
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ARGC is needed if you're using optional argument, to verify whether or
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not an argument has been passed, else it's possible to use a leaked ARGn
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from Gosub (app_stack). So now you can safely do
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${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.
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res_srtp
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------------------
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* SRTP replay protection has been added to res_srtp and
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a new configuration option "srtpreplayprotection" has
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been added to the rtp.conf config file. For security
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reasons, the default setting is "yes". Buggy clients
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may not handle this correctly which could result in
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no, or one way, audio and Asterisk error messages like
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"replay check failed".
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 18.1.0 to Asterisk 18.2.0 ------------
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------------------------------------------------------------------------------
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14
UPGRADE.txt
14
UPGRADE.txt
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@ -18,6 +18,20 @@
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===
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===========================================================
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 18.2.2 to Asterisk 18.3.0 ------------
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------------------------------------------------------------------------------
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res_srtp
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------------------
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* SRTP replay protection has been added to res_srtp and
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a new configuration option "srtpreplayprotection" has
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been added to the rtp.conf config file. For security
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reasons, the default setting is "yes". Buggy clients
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may not handle this correctly which could result in
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no, or one way, audio and Asterisk error messages like
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"replay check failed".
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------------------------------------------------------------------------------
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--- New functionality introduced in Asterisk 18.0.0 --------------------------
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------------------------------------------------------------------------------
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@ -1,6 +0,0 @@
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Subject: chan_pjsip, app_transfer
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Added TRANSFERSTATUSPROTOCOL variable. When transfer is performed,
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transfers can pass a protocol specific error code.
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Example, in SIP 3xx-6xx represent any SIP specific error received when
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performing a REFER.
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@ -1,4 +0,0 @@
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Subject: chan_iax2
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You can now specify a default "auth" method in the
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[general] section of iax.conf
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@ -1,20 +0,0 @@
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Subject: func_odbc
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Introduce an ARGC variable for func_odbc functions, along with a minargs
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per-function configuration option.
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minargs enables enforcing of minimum count of arguments to pass to
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func_odbc, so if you're unconditionally using ARG1 through ARG4 then
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this should be set to 4. func_odbc will generate an error in this case,
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so for example
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[FOO]
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minargs = 4
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and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
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potentially leaked ARG4 from Gosub().
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ARGC is needed if you're using optional argument, to verify whether or
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not an argument has been passed, else it's possible to use a leaked ARGn
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from Gosub (app_stack). So now you can safely do
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${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.
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@ -1,5 +0,0 @@
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Subject: app_mixmonitor
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app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and
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MixMonitorMute when the channel monitoring is started, stopped and muted (or
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unmuted) respectively.
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@ -1,9 +0,0 @@
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Subject: res_srtp
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SRTP replay protection has been added to res_srtp and
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a new configuration option "srtpreplayprotection" has
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been added to the rtp.conf config file. For security
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reasons, the default setting is "yes". Buggy clients
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may not handle this correctly which could result in
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no, or one way, audio and Asterisk error messages like
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"replay check failed".
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@ -1,9 +0,0 @@
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Subject: res_srtp
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SRTP replay protection has been added to res_srtp and
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a new configuration option "srtpreplayprotection" has
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been added to the rtp.conf config file. For security
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reasons, the default setting is "yes". Buggy clients
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may not handle this correctly which could result in
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no, or one way, audio and Asterisk error messages like
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"replay check failed".
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