diff --git a/CHANGES b/CHANGES index b00c4b4777..b59886a917 100644 --- a/CHANGES +++ b/CHANGES @@ -12,6 +12,137 @@ === ============================================================================== +------------------------------------------------------------------------------ +--- New functionality introduced in Asterisk 17.0.0 -------------------------- +------------------------------------------------------------------------------ + +Bridging +------------------ + * The bridging core no longer uses the stasis cache for bridge + snapshots. The latest bridge snapshot is now stored on the + ast_bridge structure itself. + + The following APIs are no longer available since the stasis cache + is no longer used: + ast_bridge_topic_cached() + ast_bridge_topic_all_cached() + + A topic pool is now used for individual bridge topics. + + The ast_bridge_cache() function was removed since there's no + longer a separate container of snapshots. + + A new function "ast_bridges()" was created to retrieve the + container of all bridges. Users formerly calling + ast_bridge_cache() can use the new function to iterate over + bridges and retrieve the latest snapshot directly from the + bridge. + + The ast_bridge_snapshot_get_latest() function was renamed to + ast_bridge_get_snapshot_by_uniqueid(). + + A new function "ast_bridge_get_snapshot()" was created to retrieve + the bridge snapshot directly from the bridge structure. + + The ast_bridge_topic_all() function now returns a normal topic + not a cached one so you can't use stasis cache functions on it + either. + + The ast_bridge_snapshot_type() stasis message now has the + ast_bridge_snapshot_update structure as it's data. It contains + the last snapshot and the new one. + +Channels +------------------ + * The core no longer uses the stasis cache for channels snapshots. + The following APIs are no longer available: + ast_channel_topic_cached() + ast_channel_topic_all_cached() + The ast_channel_cache_all() and ast_channel_cache_by_name() functions + now returns an ao2_container of ast_channel_snapshots rather than a + container of stasis_messages therefore you can't call stasis_cache + functions on it. + The ast_channel_topic_all() function now returns a normal topic, + not a cached one so you can't use stasis cache functions on it either. + The ast_channel_snapshot_type() stasis message now has the + ast_channel_snapshot_update structure as it's data. + ast_channel_snapshot_get_latest() still returns the latest snapshot. + +chan_sip +------------------ + * The chan_sip module is now deprecated, users should migrate to the + replacement module chan_pjsip. See guides at the Asterisk Wiki: + https://wiki.asterisk.org/wiki/x/tAHOAQ + https://wiki.asterisk.org/wiki/x/hYCLAQ + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 16.0.0 to Asterisk 17.0.0 ------------ +------------------------------------------------------------------------------ + +AttendedTransfer +------------------ + * A new application, this will queue up attended transfer to the given extension. + +BlindTransfer +------------------ + * A new application, this will redirect all channels currently + bridged to the caller channel to the specified destination. + +ConfBridge +------------------ + * Add "average_all", "highest_all", and "lowest_all" values for + the remb_behavior option. These values operate on a bridge + level instead of a per-source level. This means that a single + REMB value is calculated and sent to every sender, instead of + a REMB value that is unique for the specific sender.. + +Dial +------------------ + * Add RINGTIME and RINGTIME_MS variables containing respectively seconds and + milliseconds between creation of the dialing channel and receiving the first + RINGING signal + + Add PROGRESSTIME and PROGRESSTIME_MS variables analogous to the above with respect to + the PROGRESS signal. Shorter of these two times should be equivalent to + the PDD (Post Dial Delay) value + + Add DIALEDTIME_MS and ANSWEREDTIME_MS variables to get millisecond resolution + versions of DIALEDTIME and ANSWEREDTIME + +RTP/ICE +------------------ + * You can now indicate that you'd like an ice_host_candidate's local address + to be published as well as the mapped address. See the sample rtp.conf + for more information. + +ReadExten +------------------ + * Add 'p' option to stop reading extension if user presses '#' key. + +pbx_dundi +------------------ + * The DUNDi PBX module now supports IPv4/IPv6 dual binding. + +res_pjsip +------------------ + * Added a new PJSIP global setting called norefersub. + Default is true to keep support working as before. + + res_pjsip_refer configures PJSIP norefersub capability accordingly. + + Checks the PJSIP global setting value. + If it is true (default) it adds the norefersub capability to PJSIP. + If it is false (disabled) it does not add the norefersub capability + to PJSIP. + + This is useful for Cisco switches that do not follow RFC4488. + +res_rtp_asterisk +------------------ + * DTLS packets will now be fragmented according to the MTU as set in rtp.conf. This + allows larger certificates to be used for the DTLS negotiation. By default this value + is 1200. + ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 16.2.0 to Asterisk 16.3.0 ---------- ------------------------------------------------------------------------------ diff --git a/UPGRADE.txt b/UPGRADE.txt index 3abdb34ce2..68a9e09f2a 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -18,6 +18,118 @@ === =========================================================== +------------------------------------------------------------------------------ +--- New functionality introduced in Asterisk 17.0.0 -------------------------- +------------------------------------------------------------------------------ + +Applications +------------------ + * The JabberStatus application, deprecated in Asterisk 12, has been removed. + +Bridging +------------------ + * The bridging core no longer uses the stasis cache for bridge + snapshots. The latest bridge snapshot is now stored on the + ast_bridge structure itself. + + The following APIs are no longer available since the stasis cache + is no longer used: + ast_bridge_topic_cached() + ast_bridge_topic_all_cached() + + A topic pool is now used for individual bridge topics. + + The ast_bridge_cache() function was removed since there's no + longer a separate container of snapshots. + + A new function "ast_bridges()" was created to retrieve the + container of all bridges. Users formerly calling + ast_bridge_cache() can use the new function to iterate over + bridges and retrieve the latest snapshot directly from the + bridge. + + The ast_bridge_snapshot_get_latest() function was renamed to + ast_bridge_get_snapshot_by_uniqueid(). + + A new function "ast_bridge_get_snapshot()" was created to retrieve + the bridge snapshot directly from the bridge structure. + + The ast_bridge_topic_all() function now returns a normal topic + not a cached one so you can't use stasis cache functions on it + either. + + The ast_bridge_snapshot_type() stasis message now has the + ast_bridge_snapshot_update structure as it's data. It contains + the last snapshot and the new one. + +Build +------------------ + * Asterisk headers are no longer installed and uninstalled automatically when + performing a "make install" or a "make uninstall". To install/uninstall the + headers, use "make install-headers" and "make uninstall-headers". The headers + also continue to be uninstalled when performing a "make uninstall-all". + +Channels +------------------ + * The core no longer uses the stasis cache for channels snapshots. + The following APIs are no longer available: + ast_channel_topic_cached() + ast_channel_topic_all_cached() + The ast_channel_cache_all() and ast_channel_cache_by_name() functions + now returns an ao2_container of ast_channel_snapshots rather than a + container of stasis_messages therefore you can't call stasis_cache + functions on it. + The ast_channel_topic_all() function now returns a normal topic, + not a cached one so you can't use stasis cache functions on it either. + The ast_channel_snapshot_type() stasis message now has the + ast_channel_snapshot_update structure as it's data. + ast_channel_snapshot_get_latest() still returns the latest snapshot. + +chan_sip +------------------ + * The chan_sip module is now deprecated, users should migrate to the + replacement module chan_pjsip. See guides at the Asterisk Wiki: + https://wiki.asterisk.org/wiki/x/tAHOAQ + https://wiki.asterisk.org/wiki/x/hYCLAQ + +func_callerid +------------------ + * The CALLERPRES() dialplan function, deprecated in Asterisk 1.8, has been + removed. + +res_parking +------------------ + * The PARKINGSLOT channel variable, deprecated in Asterisk 12 in favor of the + PARKING_SPACE channel variable, will no longer be set. + +res_xmpp +------------------ + * The JabberStatus application, deprecated in Asterisk 12, has been removed. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 16.0.0 to Asterisk 17.0.0 ------------ +------------------------------------------------------------------------------ + +Core +------------------ + * res_pjsip_pubsub is now required so call transfer progress can be monitored + and reported in the channel variable TRANSFERSTATUS. + +app_voicemail.c +------------------ + * The "Voicemail Build Options" section of menuselect has been removed along with + the FILE_STORAGE, ODBC_STORAGE and IMAP_STORAGE menuselect options. All 3 variants + of the voicemail app can now be built at the same by enabling app_voicemail, + app_voicemail_imap, and app_voicemail_odbc under the "Applications" section. + By default, only app_voicemail is enabled. Also, the modules.conf sample has + been updated to "noload" app_voicemail_imap and app_voicemail_odbc should they + all be built. Packagers must update their build scripts appropriately. + +chan_pjsip +------------------ + * res_pjsip_pubsub is now required so call transfer progress can be monitored + and reported in the channel variable TRANSFERSTATUS. + New in 16.0.0: app_fax: diff --git a/doc/CHANGES-staging/app_attended_transfer.txt b/doc/CHANGES-staging/app_attended_transfer.txt deleted file mode 100644 index 97929e4b08..0000000000 --- a/doc/CHANGES-staging/app_attended_transfer.txt +++ /dev/null @@ -1,3 +0,0 @@ -Subject: AttendedTransfer - -A new application, this will queue up attended transfer to the given extension. diff --git a/doc/CHANGES-staging/app_blind_transfer.txt b/doc/CHANGES-staging/app_blind_transfer.txt deleted file mode 100644 index dc86df0031..0000000000 --- a/doc/CHANGES-staging/app_blind_transfer.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: BlindTransfer - -A new application, this will redirect all channels currently -bridged to the caller channel to the specified destination. diff --git a/doc/CHANGES-staging/app_confbridge_remb_behavior_all.txt b/doc/CHANGES-staging/app_confbridge_remb_behavior_all.txt deleted file mode 100644 index 6110a6f1b8..0000000000 --- a/doc/CHANGES-staging/app_confbridge_remb_behavior_all.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: ConfBridge - -Add "average_all", "highest_all", and "lowest_all" values for -the remb_behavior option. These values operate on a bridge -level instead of a per-source level. This means that a single -REMB value is calculated and sent to every sender, instead of -a REMB value that is unique for the specific sender.. diff --git a/doc/CHANGES-staging/app_dial_ringtime_progresstime.txt b/doc/CHANGES-staging/app_dial_ringtime_progresstime.txt deleted file mode 100644 index 9b5cdd5089..0000000000 --- a/doc/CHANGES-staging/app_dial_ringtime_progresstime.txt +++ /dev/null @@ -1,12 +0,0 @@ -Subject: Dial - -Add RINGTIME and RINGTIME_MS variables containing respectively seconds and -milliseconds between creation of the dialing channel and receiving the first -RINGING signal - -Add PROGRESSTIME and PROGRESSTIME_MS variables analogous to the above with respect to -the PROGRESS signal. Shorter of these two times should be equivalent to -the PDD (Post Dial Delay) value - -Add DIALEDTIME_MS and ANSWEREDTIME_MS variables to get millisecond resolution -versions of DIALEDTIME and ANSWEREDTIME diff --git a/doc/CHANGES-staging/app_readexten_pound.txt b/doc/CHANGES-staging/app_readexten_pound.txt deleted file mode 100644 index 551f751d10..0000000000 --- a/doc/CHANGES-staging/app_readexten_pound.txt +++ /dev/null @@ -1,3 +0,0 @@ -Subject: ReadExten - -Add 'p' option to stop reading extension if user presses '#' key. diff --git a/doc/CHANGES-staging/bridging_stasis_cache.txt b/doc/CHANGES-staging/bridging_stasis_cache.txt deleted file mode 100644 index df6b3cd103..0000000000 --- a/doc/CHANGES-staging/bridging_stasis_cache.txt +++ /dev/null @@ -1,36 +0,0 @@ -Subject: Bridging -Master-Only: true - -The bridging core no longer uses the stasis cache for bridge -snapshots. The latest bridge snapshot is now stored on the -ast_bridge structure itself. - -The following APIs are no longer available since the stasis cache -is no longer used: - ast_bridge_topic_cached() - ast_bridge_topic_all_cached() - -A topic pool is now used for individual bridge topics. - -The ast_bridge_cache() function was removed since there's no -longer a separate container of snapshots. - -A new function "ast_bridges()" was created to retrieve the -container of all bridges. Users formerly calling -ast_bridge_cache() can use the new function to iterate over -bridges and retrieve the latest snapshot directly from the -bridge. - -The ast_bridge_snapshot_get_latest() function was renamed to -ast_bridge_get_snapshot_by_uniqueid(). - -A new function "ast_bridge_get_snapshot()" was created to retrieve -the bridge snapshot directly from the bridge structure. - -The ast_bridge_topic_all() function now returns a normal topic -not a cached one so you can't use stasis cache functions on it -either. - -The ast_bridge_snapshot_type() stasis message now has the -ast_bridge_snapshot_update structure as it's data. It contains -the last snapshot and the new one. diff --git a/doc/CHANGES-staging/chan_sip_deprecated.txt b/doc/CHANGES-staging/chan_sip_deprecated.txt deleted file mode 100644 index cffd1db565..0000000000 --- a/doc/CHANGES-staging/chan_sip_deprecated.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: chan_sip -Master-Only: true - -The chan_sip module is now deprecated, users should migrate to the -replacement module chan_pjsip. See guides at the Asterisk Wiki: - https://wiki.asterisk.org/wiki/x/tAHOAQ - https://wiki.asterisk.org/wiki/x/hYCLAQ diff --git a/doc/CHANGES-staging/channels_stasis_cache.txt b/doc/CHANGES-staging/channels_stasis_cache.txt deleted file mode 100644 index b4dbfc3446..0000000000 --- a/doc/CHANGES-staging/channels_stasis_cache.txt +++ /dev/null @@ -1,16 +0,0 @@ -Subject: Channels -Master-Only: true - -The core no longer uses the stasis cache for channels snapshots. -The following APIs are no longer available: - ast_channel_topic_cached() - ast_channel_topic_all_cached() -The ast_channel_cache_all() and ast_channel_cache_by_name() functions -now returns an ao2_container of ast_channel_snapshots rather than a -container of stasis_messages therefore you can't call stasis_cache -functions on it. -The ast_channel_topic_all() function now returns a normal topic, -not a cached one so you can't use stasis cache functions on it either. -The ast_channel_snapshot_type() stasis message now has the -ast_channel_snapshot_update structure as it's data. -ast_channel_snapshot_get_latest() still returns the latest snapshot. diff --git a/doc/CHANGES-staging/pbx_dundi_ipv6.txt b/doc/CHANGES-staging/pbx_dundi_ipv6.txt deleted file mode 100644 index c15ae44979..0000000000 --- a/doc/CHANGES-staging/pbx_dundi_ipv6.txt +++ /dev/null @@ -1,3 +0,0 @@ -Subject: pbx_dundi - -The DUNDi PBX module now supports IPv4/IPv6 dual binding. diff --git a/doc/CHANGES-staging/res_pjsip_add_norefersub_global_config.txt b/doc/CHANGES-staging/res_pjsip_add_norefersub_global_config.txt deleted file mode 100644 index e0573bc250..0000000000 --- a/doc/CHANGES-staging/res_pjsip_add_norefersub_global_config.txt +++ /dev/null @@ -1,13 +0,0 @@ -Subject: res_pjsip - -Added a new PJSIP global setting called norefersub. -Default is true to keep support working as before. - -res_pjsip_refer configures PJSIP norefersub capability accordingly. - -Checks the PJSIP global setting value. -If it is true (default) it adds the norefersub capability to PJSIP. -If it is false (disabled) it does not add the norefersub capability -to PJSIP. - -This is useful for Cisco switches that do not follow RFC4488. diff --git a/doc/CHANGES-staging/res_rtp_asterisk_dtls_fragmentation.txt b/doc/CHANGES-staging/res_rtp_asterisk_dtls_fragmentation.txt deleted file mode 100644 index dfc5984d6f..0000000000 --- a/doc/CHANGES-staging/res_rtp_asterisk_dtls_fragmentation.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_rtp_asterisk - -DTLS packets will now be fragmented according to the MTU as set in rtp.conf. This -allows larger certificates to be used for the DTLS negotiation. By default this value -is 1200. diff --git a/doc/CHANGES-staging/rtp_ice_include_local_address.txt b/doc/CHANGES-staging/rtp_ice_include_local_address.txt deleted file mode 100644 index e5a65e5f86..0000000000 --- a/doc/CHANGES-staging/rtp_ice_include_local_address.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: RTP/ICE - -You can now indicate that you'd like an ice_host_candidate's local address -to be published as well as the mapped address. See the sample rtp.conf -for more information. diff --git a/doc/UPGRADE-staging/Build_all_3_app_voicemail_variants_at_the_same_time.txt b/doc/UPGRADE-staging/Build_all_3_app_voicemail_variants_at_the_same_time.txt deleted file mode 100644 index 22fb4f7fa5..0000000000 --- a/doc/UPGRADE-staging/Build_all_3_app_voicemail_variants_at_the_same_time.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: app_voicemail.c - -The "Voicemail Build Options" section of menuselect has been removed along with -the FILE_STORAGE, ODBC_STORAGE and IMAP_STORAGE menuselect options. All 3 variants -of the voicemail app can now be built at the same by enabling app_voicemail, -app_voicemail_imap, and app_voicemail_odbc under the "Applications" section. -By default, only app_voicemail is enabled. Also, the modules.conf sample has -been updated to "noload" app_voicemail_imap and app_voicemail_odbc should they -all be built. Packagers must update their build scripts appropriately. diff --git a/doc/UPGRADE-staging/applications_jabberstatus.txt b/doc/UPGRADE-staging/applications_jabberstatus.txt deleted file mode 100644 index 93e411297c..0000000000 --- a/doc/UPGRADE-staging/applications_jabberstatus.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: Applications -Master-Only: true - -The JabberStatus application, deprecated in Asterisk 12, has been removed. diff --git a/doc/UPGRADE-staging/bridging_stasis_cache.txt b/doc/UPGRADE-staging/bridging_stasis_cache.txt deleted file mode 100644 index df6b3cd103..0000000000 --- a/doc/UPGRADE-staging/bridging_stasis_cache.txt +++ /dev/null @@ -1,36 +0,0 @@ -Subject: Bridging -Master-Only: true - -The bridging core no longer uses the stasis cache for bridge -snapshots. The latest bridge snapshot is now stored on the -ast_bridge structure itself. - -The following APIs are no longer available since the stasis cache -is no longer used: - ast_bridge_topic_cached() - ast_bridge_topic_all_cached() - -A topic pool is now used for individual bridge topics. - -The ast_bridge_cache() function was removed since there's no -longer a separate container of snapshots. - -A new function "ast_bridges()" was created to retrieve the -container of all bridges. Users formerly calling -ast_bridge_cache() can use the new function to iterate over -bridges and retrieve the latest snapshot directly from the -bridge. - -The ast_bridge_snapshot_get_latest() function was renamed to -ast_bridge_get_snapshot_by_uniqueid(). - -A new function "ast_bridge_get_snapshot()" was created to retrieve -the bridge snapshot directly from the bridge structure. - -The ast_bridge_topic_all() function now returns a normal topic -not a cached one so you can't use stasis cache functions on it -either. - -The ast_bridge_snapshot_type() stasis message now has the -ast_bridge_snapshot_update structure as it's data. It contains -the last snapshot and the new one. diff --git a/doc/UPGRADE-staging/chan_pjsip_refer_fix.txt b/doc/UPGRADE-staging/chan_pjsip_refer_fix.txt deleted file mode 100644 index 301930f8bd..0000000000 --- a/doc/UPGRADE-staging/chan_pjsip_refer_fix.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: chan_pjsip -Subject: Core - -res_pjsip_pubsub is now required so call transfer progress can be monitored -and reported in the channel variable TRANSFERSTATUS. diff --git a/doc/UPGRADE-staging/chan_sip_deprecated.txt b/doc/UPGRADE-staging/chan_sip_deprecated.txt deleted file mode 100644 index cffd1db565..0000000000 --- a/doc/UPGRADE-staging/chan_sip_deprecated.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: chan_sip -Master-Only: true - -The chan_sip module is now deprecated, users should migrate to the -replacement module chan_pjsip. See guides at the Asterisk Wiki: - https://wiki.asterisk.org/wiki/x/tAHOAQ - https://wiki.asterisk.org/wiki/x/hYCLAQ diff --git a/doc/UPGRADE-staging/channels_stasis_cache.txt b/doc/UPGRADE-staging/channels_stasis_cache.txt deleted file mode 100644 index b4dbfc3446..0000000000 --- a/doc/UPGRADE-staging/channels_stasis_cache.txt +++ /dev/null @@ -1,16 +0,0 @@ -Subject: Channels -Master-Only: true - -The core no longer uses the stasis cache for channels snapshots. -The following APIs are no longer available: - ast_channel_topic_cached() - ast_channel_topic_all_cached() -The ast_channel_cache_all() and ast_channel_cache_by_name() functions -now returns an ao2_container of ast_channel_snapshots rather than a -container of stasis_messages therefore you can't call stasis_cache -functions on it. -The ast_channel_topic_all() function now returns a normal topic, -not a cached one so you can't use stasis cache functions on it either. -The ast_channel_snapshot_type() stasis message now has the -ast_channel_snapshot_update structure as it's data. -ast_channel_snapshot_get_latest() still returns the latest snapshot. diff --git a/doc/UPGRADE-staging/func_callerid_callerpres.txt b/doc/UPGRADE-staging/func_callerid_callerpres.txt deleted file mode 100644 index 003d67d699..0000000000 --- a/doc/UPGRADE-staging/func_callerid_callerpres.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: func_callerid -Master-Only: true - -The CALLERPRES() dialplan function, deprecated in Asterisk 1.8, has been -removed. diff --git a/doc/UPGRADE-staging/install_headers.txt b/doc/UPGRADE-staging/install_headers.txt deleted file mode 100644 index d932512a1b..0000000000 --- a/doc/UPGRADE-staging/install_headers.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: Build -Master-Only: true - -Asterisk headers are no longer installed and uninstalled automatically when -performing a "make install" or a "make uninstall". To install/uninstall the -headers, use "make install-headers" and "make uninstall-headers". The headers -also continue to be uninstalled when performing a "make uninstall-all". diff --git a/doc/UPGRADE-staging/res_parking_parkingslot.txt b/doc/UPGRADE-staging/res_parking_parkingslot.txt deleted file mode 100644 index b5b4cbc392..0000000000 --- a/doc/UPGRADE-staging/res_parking_parkingslot.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_parking -Master-Only: true - -The PARKINGSLOT channel variable, deprecated in Asterisk 12 in favor of the -PARKING_SPACE channel variable, will no longer be set. diff --git a/doc/UPGRADE-staging/res_xmpp_jabberstatus.txt b/doc/UPGRADE-staging/res_xmpp_jabberstatus.txt deleted file mode 100644 index 4400278632..0000000000 --- a/doc/UPGRADE-staging/res_xmpp_jabberstatus.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: res_xmpp -Master-Only: true - -The JabberStatus application, deprecated in Asterisk 12, has been removed.