diff --git a/CHANGES b/CHANGES index f02db0179f981f6b862a7e078cc0b751a6a257be..401a886c62a1ee554248fac2df15b4a0a65bd047 100644 --- a/CHANGES +++ b/CHANGES @@ -12,6 +12,72 @@ === ============================================================================== +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 20.1.0 to Asterisk 20.2.0 ------------ +------------------------------------------------------------------------------ + +app_broadcast +------------------ + * A Broadcast application is now available which allows + for asynchronous one-to-many and many-to-one channel audio. + +app_directory +------------------ + * A new option 's' has been added to the Directory() application that + will skip calling the extension and instead set the extension as + DIRECTORY_EXTEN channel variable. + +app_read +------------------ + * A new option 'e' has been added to allow Read() to return the + terminator as the dialed digits in the case where only the terminator + is entered. + +app_senddtmf +------------------ + * A new option has been added to SendDTMF() which will answer the + specified channel if it is not already up. If no channel is specified, + the current channel will be answered instead. + +app_signal +------------------ + * Adds Signal and WaitForSignal applications + which can be used for signaling or as a + simple message queue in the dialplan. + +func_json +------------------ + * Additional parsing capabilities have been added to the + JSON_DECODE function, including support for arrays + and recursive indexing. + +res_phoneprov +------------------ + * On multihomed Asterisk servers with dynamic SERVER template variables, + reloading this module is no longer required when re-provisioning your + phone to another interface address (e.g. when moving between VLANs.) + +res_pjsip_rfc3326 +------------------ + * Add ability to set HANGUPCAUSE when SIP causecode received in BYE Reason header (in + addition to currently supported Q.850). The first header found will be used to set + the HANGUPCAUSE variable. + +res_pjsip_session +------------------ + * The overlap_context option now allows explicitly + specifying a context to use for overlap dialing matches. + +res_rtp_asterisk +------------------ + * This module has been updated to provide additional + quality statistics in the form of an Asterisk + Media Experience Score. The score is available using + the same mechanisms you'd use to retrieve jitter, loss, + and rtt statistics. For more information about the + score and how to retrieve it, see + https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score + ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 20.0.0 to Asterisk 20.1.0 ------------ ------------------------------------------------------------------------------ diff --git a/UPGRADE.txt b/UPGRADE.txt index 42fff46bd8bcab70c1af21e24ea3ba974902b1ac..972e4fb17a4a7e69dc55d0b4725ddcc04d46b01c 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -18,6 +18,19 @@ === =========================================================== +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 20.1.0 to Asterisk 20.2.0 ------------ +------------------------------------------------------------------------------ + +app_playback +------------------ + * In Asterisk 11, if a channel was redirected away during Playback(), + the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12 + (specifically commit 7d9871b3940fa50e85039aef6a8fb9870a7615b9) that + behavior was inadvertently changed and the same operation would result + in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11 + behavior has been restored. + ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 20.0.0 to Asterisk 20.1.0 ------------ ------------------------------------------------------------------------------ diff --git a/doc/CHANGES-staging/app_broadcast.txt b/doc/CHANGES-staging/app_broadcast.txt deleted file mode 100644 index 03e68483624ee32f343446ea8d3c4f2e57934b2d..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/app_broadcast.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: app_broadcast - -A Broadcast application is now available which allows -for asynchronous one-to-many and many-to-one channel audio. diff --git a/doc/CHANGES-staging/app_directory_skip_call.txt b/doc/CHANGES-staging/app_directory_skip_call.txt deleted file mode 100644 index 83687fe3f9c2d56e036d48980c5b1cc9781d109f..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/app_directory_skip_call.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_directory - -A new option 's' has been added to the Directory() application that -will skip calling the extension and instead set the extension as -DIRECTORY_EXTEN channel variable. diff --git a/doc/CHANGES-staging/app_read_return_terminator.txt b/doc/CHANGES-staging/app_read_return_terminator.txt deleted file mode 100644 index 2987f77ea7ca0ae7f19b18449deac4d6be480507..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/app_read_return_terminator.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_read - -A new option 'e' has been added to allow Read() to return the -terminator as the dialed digits in the case where only the terminator -is entered. diff --git a/doc/CHANGES-staging/app_senddtmf_answer.txt b/doc/CHANGES-staging/app_senddtmf_answer.txt deleted file mode 100644 index 76811e3a7fc6700be528aa6811d7f24535ac836e..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/app_senddtmf_answer.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_senddtmf - -A new option has been added to SendDTMF() which will answer the -specified channel if it is not already up. If no channel is specified, -the current channel will be answered instead. diff --git a/doc/CHANGES-staging/app_signal.txt b/doc/CHANGES-staging/app_signal.txt deleted file mode 100644 index b3b108d82101dfefbe35df119f0f75679394c8ad..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/app_signal.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_signal - -Adds Signal and WaitForSignal applications -which can be used for signaling or as a -simple message queue in the dialplan. diff --git a/doc/CHANGES-staging/func_json_additions.txt b/doc/CHANGES-staging/func_json_additions.txt deleted file mode 100644 index 963f0b18e2517bcabeb15171bd2c29b5afbc97b3..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/func_json_additions.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: func_json - -Additional parsing capabilities have been added to the -JSON_DECODE function, including support for arrays -and recursive indexing. diff --git a/doc/CHANGES-staging/res_phoneprov_multihomed_server.txt b/doc/CHANGES-staging/res_phoneprov_multihomed_server.txt deleted file mode 100644 index ff68014570731ea44d3deec1670641ec27919f71..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/res_phoneprov_multihomed_server.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_phoneprov - -On multihomed Asterisk servers with dynamic SERVER template variables, -reloading this module is no longer required when re-provisioning your -phone to another interface address (e.g. when moving between VLANs.) diff --git a/doc/CHANGES-staging/res_pjsip_session_overlap.txt b/doc/CHANGES-staging/res_pjsip_session_overlap.txt deleted file mode 100644 index 5523f3c08666425dbc5a9b0f1ee41405ad1fa7a7..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/res_pjsip_session_overlap.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: res_pjsip_session - -The overlap_context option now allows explicitly -specifying a context to use for overlap dialing matches. diff --git a/doc/CHANGES-staging/res_rtp_asterisk.txt b/doc/CHANGES-staging/res_rtp_asterisk.txt deleted file mode 100644 index 9c8e05f0b6da5642583df8153139814667afa65f..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/res_rtp_asterisk.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: res_rtp_asterisk - -This module has been updated to provide additional -quality statistics in the form of an Asterisk -Media Experience Score. The score is available using -the same mechanisms you'd use to retrieve jitter, loss, -and rtt statistics. For more information about the -score and how to retrieve it, see -https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score diff --git a/doc/CHANGES-staging/res_rtp_rfc3326_sip.txt b/doc/CHANGES-staging/res_rtp_rfc3326_sip.txt deleted file mode 100644 index 62a73925ca1afeac8aca4c76870e58ac21265856..0000000000000000000000000000000000000000 --- a/doc/CHANGES-staging/res_rtp_rfc3326_sip.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_pjsip_rfc3326 - -Add ability to set HANGUPCAUSE when SIP causecode received in BYE Reason header (in -addition to currently supported Q.850). The first header found will be used to set -the HANGUPCAUSE variable. diff --git a/doc/UPGRADE-staging/app_playback_playbackstatus.txt b/doc/UPGRADE-staging/app_playback_playbackstatus.txt deleted file mode 100644 index 49302b79668a3797329f06dc24eb49ab10f3e741..0000000000000000000000000000000000000000 --- a/doc/UPGRADE-staging/app_playback_playbackstatus.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: app_playback - -In Asterisk 11, if a channel was redirected away during Playback(), -the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12 -(specifically commit 7d9871b3940fa50e85039aef6a8fb9870a7615b9) that -behavior was inadvertently changed and the same operation would result -in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11 -behavior has been restored.