diff --git a/CHANGES b/CHANGES index a0b4eccdc37a0986c8776c3c086687d57660e0dd..cf43ed4799797fc94700bf67a9343797a411762f 100644 --- a/CHANGES +++ b/CHANGES @@ -12,6 +12,107 @@ === ============================================================================== +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 19.4.0 to Asterisk 19.5.0 ------------ +------------------------------------------------------------------------------ + +app_confbridge +------------------ + * Added the hear_own_join_sound option to the confbridge user profile to + control who hears the sound_join audio file. When set to 'yes' the user + entering the conference and the participants already in the conference + will hear the sound_join audio file. When set to 'no' the user entering + the conference will not hear the sound_join audio file, but the + participants already in the conference will hear the sound_join audio file. + + * Adds the CONFBRIDGE_CHANNELS function which can + be used to retrieve a list of channels in a ConfBridge, + optionally filtered by a particular category. This + list can then be used with functions like SHIFT, POP, + UNSHIFT, etc. + +app_queue +------------------ + * The m option now allows an override music on hold + class to be specified for the Queue application + within the dialplan. + +app_voicemail +------------------ + * The r option has been added, which prevents deletion + of messages from VoiceMailMain, which can be + useful for shared mailboxes. + +ari +------------------ + * Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP) + to ARI channel resources as 'protocol_id'. + + ASTERISK-30027 + +chan_dahdi +------------------ + * Previously, cadences were appended on dahdi restart, + rather than reloaded. This prevented cadences from + being updated and maxed out the available cadences + if reloaded multiple times. This behavior is fixed + so that reloading cadences is idempotent and cadences + can actually be reloaded. + +chan_pjsip +------------------ + * added global config option "allow_sending_180_after_183" + + Allow Asterisk to send 180 Ringing to an endpoint + after 183 Session Progress has been send. + If disabled Asterisk will instead send only a + 183 Session Progress to the endpoint. + + * Hook flash events can now be sent on a PJSIP channel + if requested to do so. + +chan_sip +------------------ + * Session timers get removed on UPDATE + Fix if Asterisk receives a SIP REFER with Session-Timers UAC + that Asterisk maintains Session-Timers when sending UPDATE request + +cli +------------------ + * A new CLI command 'dialplan eval function' has been + added which allows users to test the behavior of + dialplan function calls directly from the CLI. + +func_db +------------------ + * The function DB_KEYCOUNT has been added, which + returns the cardinality of the keys at a specified + prefix in AstDB, i.e. the number of keys at a + given prefix. + +func_evalexten +------------------ + * This adds the EVAL_EXTEN function which may be + used to evaluate data at dialplan extensions. + +res_agi +------------------ + * Agi command 'exec' can now be enabled + to evaluate dialplan functions and variables + by setting the variable AGIEXECFULL to yes. + +res_parking +------------------ + * An m option to Park and ParkAndAnnounce now allows + specifying a music on hold class override. + +stasis_channels +------------------ + * Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP) + to ARI channel resources as 'protocol_id'. + + ASTERISK-30027 + ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 19.3.1 to Asterisk 19.3.2 ------------ ------------------------------------------------------------------------------ diff --git a/UPGRADE.txt b/UPGRADE.txt index fedcff9cf50afeddcc4eed4dd7b78e7446f7d7b5..bfa9f9eb684f8ce9c82d4ee4faa7fa17e1e0f7f0 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -18,6 +18,18 @@ === =========================================================== +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 19.4.0 to Asterisk 19.5.0 ------------ +------------------------------------------------------------------------------ + +res_pjsip +------------------ + * The 'async_operations' setting on transports is no longer + obeyed and instead is always set to 1. This is due to the + functionality not being applicable to Asterisk and causing + excess unnecessary memory usage. This setting will now be + ignored but can also be removed from the configuration file. + ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 19.2.0 to Asterisk 19.3.0 ------------ ------------------------------------------------------------------------------ diff --git a/doc/CHANGES-staging/app_confbridge_channels.txt b/doc/CHANGES-staging/app_confbridge_channels.txt deleted file mode 100644 index 485f6642682c413419c03fe6f438de2f8d85b5c9..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/app_confbridge_channels.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: app_confbridge - -Adds the CONFBRIDGE_CHANNELS function which can -be used to retrieve a list of channels in a ConfBridge, -optionally filtered by a particular category. This -list can then be used with functions like SHIFT, POP, -UNSHIFT, etc. diff --git a/doc/CHANGES-staging/app_confbridge_hear_join.txt b/doc/CHANGES-staging/app_confbridge_hear_join.txt deleted file mode 100644 index 40f23836ff32d4fd91dc58a533aac25f2223c8f8..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/app_confbridge_hear_join.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: app_confbridge - -Added the hear_own_join_sound option to the confbridge user profile to -control who hears the sound_join audio file. When set to 'yes' the user -entering the conference and the participants already in the conference -will hear the sound_join audio file. When set to 'no' the user entering -the conference will not hear the sound_join audio file, but the -participants already in the conference will hear the sound_join audio file. diff --git a/doc/CHANGES-staging/app_queue_music.txt b/doc/CHANGES-staging/app_queue_music.txt deleted file mode 100644 index 254a45db4560c6273b0855d8c95677385d5c1365..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/app_queue_music.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_queue - -The m option now allows an override music on hold -class to be specified for the Queue application -within the dialplan. diff --git a/doc/CHANGES-staging/app_voicemail_nodelete.txt b/doc/CHANGES-staging/app_voicemail_nodelete.txt deleted file mode 100644 index ef9589652d08174748f11e87e0a67feb5ea9a795..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/app_voicemail_nodelete.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_voicemail - -The r option has been added, which prevents deletion -of messages from VoiceMailMain, which can be -useful for shared mailboxes. diff --git a/doc/CHANGES-staging/ari_add_pvt_id_to_channel_resource.txt b/doc/CHANGES-staging/ari_add_pvt_id_to_channel_resource.txt deleted file mode 100644 index a4f008f967f4f36f4ce8735837abd44a4f544a24..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/ari_add_pvt_id_to_channel_resource.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: ari -Subject: stasis_channels - -Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP) -to ARI channel resources as 'protocol_id'. - -ASTERISK-30027 diff --git a/doc/CHANGES-staging/chan_dahdi_cadences.txt b/doc/CHANGES-staging/chan_dahdi_cadences.txt deleted file mode 100644 index b888926eee92e170980a102cb29f9ee9cc30673a..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/chan_dahdi_cadences.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: chan_dahdi - -Previously, cadences were appended on dahdi restart, -rather than reloaded. This prevented cadences from -being updated and maxed out the available cadences -if reloaded multiple times. This behavior is fixed -so that reloading cadences is idempotent and cadences -can actually be reloaded. diff --git a/doc/CHANGES-staging/chan_pjsip_180_sdp.txt b/doc/CHANGES-staging/chan_pjsip_180_sdp.txt deleted file mode 100644 index ffd14af10c28f69db09b7d29c91d069002312dc8..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/chan_pjsip_180_sdp.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: chan_pjsip - -added global config option "allow_sending_180_after_183" - -Allow Asterisk to send 180 Ringing to an endpoint -after 183 Session Progress has been send. -If disabled Asterisk will instead send only a -183 Session Progress to the endpoint. diff --git a/doc/CHANGES-staging/chan_pjsip_flash.txt b/doc/CHANGES-staging/chan_pjsip_flash.txt deleted file mode 100644 index 34da79685700ecc28d673e23eedad93a1517699d..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/chan_pjsip_flash.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: chan_pjsip - -Hook flash events can now be sent on a PJSIP channel -if requested to do so. diff --git a/doc/CHANGES-staging/chan_sip_session-timer_on_update.txt b/doc/CHANGES-staging/chan_sip_session-timer_on_update.txt deleted file mode 100644 index 259782f518d94892b70f5c6d5a9bbf5bd13dbb94..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/chan_sip_session-timer_on_update.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: chan_sip - -Session timers get removed on UPDATE -Fix if Asterisk receives a SIP REFER with Session-Timers UAC -that Asterisk maintains Session-Timers when sending UPDATE request - diff --git a/doc/CHANGES-staging/cli_eval_function.txt b/doc/CHANGES-staging/cli_eval_function.txt deleted file mode 100644 index 9f7873c738c930e0facb039aad4467a84f39b322..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/cli_eval_function.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: cli - -A new CLI command 'dialplan eval function' has been -added which allows users to test the behavior of -dialplan function calls directly from the CLI. diff --git a/doc/CHANGES-staging/func_db.txt b/doc/CHANGES-staging/func_db.txt deleted file mode 100644 index 72e333a54776ae271bbbf6c210ee47692ef90659..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/func_db.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: func_db - -The function DB_KEYCOUNT has been added, which -returns the cardinality of the keys at a specified -prefix in AstDB, i.e. the number of keys at a -given prefix. diff --git a/doc/CHANGES-staging/func_evalexten.txt b/doc/CHANGES-staging/func_evalexten.txt deleted file mode 100644 index f912bbeb5f8a74e656f773adeaaf75a853f7bf4b..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/func_evalexten.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: func_evalexten - -This adds the EVAL_EXTEN function which may be -used to evaluate data at dialplan extensions. diff --git a/doc/CHANGES-staging/res_agi.txt b/doc/CHANGES-staging/res_agi.txt deleted file mode 100644 index eb6132d614758d9c03f40bdae8a39cb5617d330c..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/res_agi.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_agi - -Agi command 'exec' can now be enabled -to evaluate dialplan functions and variables -by setting the variable AGIEXECFULL to yes. \ No newline at end of file diff --git a/doc/CHANGES-staging/res_parking_moh.txt b/doc/CHANGES-staging/res_parking_moh.txt deleted file mode 100644 index 50f589ca4390499d9c98c728bc6b020423d8860b..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/res_parking_moh.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: res_parking - -An m option to Park and ParkAndAnnounce now allows -specifying a music on hold class override. diff --git a/doc/UPGRADE-staging/res_pjsip_async_operations.txt b/doc/UPGRADE-staging/res_pjsip_async_operations.txt deleted file mode 100644 index cf9f9426dab55917ea276de9489d43ebc0c6e91d..0000000000000000000000000000000000000000 --- a/doc/UPGRADE-staging/res_pjsip_async_operations.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: res_pjsip - -The 'async_operations' setting on transports is no longer -obeyed and instead is always set to 1. This is due to the -functionality not being applicable to Asterisk and causing -excess unnecessary memory usage. This setting will now be -ignored but can also be removed from the configuration file.