<p>Benjamin Keith Ford has uploaded this change for <strong>review</strong>.</p><p><a href="https://gerrit.asterisk.org/10945">View Change</a></p><pre style="font-family: monospace,monospace; white-space: pre-wrap;">build: Revise CHANGES and UPGRADE.txt handling.<br><br>This changes the way that we handle adding changes to CHANGES and<br>UPGRADE.txt. The reason for this is because whenever someone needed to<br>make a changes to one of these files and someone else had already done<br>so, you would run into merge conflicts. With this new setup, there will<br>never be merge conflicts since all changes will be documented in the<br>doc/<file>-staging directory. The release script is now responsible for<br>merging all of these changes into the appropriate files.<br><br>There is a special format that these files have to follow in order to be<br>parsed. The files do not need to have a meaningful name, but it is<br>strongly recommended. For example, if you made a change to pjsip, you<br>may have something like this "res_pjsip_relative_title", where<br>"relative_title" is something more descriptive than that. Inside each<br>file, you will need a subject line for your change, followed by a<br>description. There can be multiple subject lines. The file may look<br>something like this:<br><br> Subject: res_pjsip<br><br> A description that explains the changes made and why. The release<br> script will handle the bulleting and section separators!<br><br> You can still separate with new lines within your<br> description.<br><br> Subject: res_pjsip<br><br> You can have more than one subject!<br><br> Subject: Core<br><br> The subjects don't have to be the same.<br><br>The headers (Subject:) are case sensative.<br><br>For more information, check out the wiki page:<br>https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt<br><br>Change-Id: I19cf4b569321c88155a65e9b0b80f6d58075dd47<br>---<br>M CHANGES<br>D UPGRADE.txt<br>A doc/CHANGES-staging/README<br>A doc/CHANGES-staging/app_voicemail_aliasescontext<br>A doc/CHANGES-staging/bridge_snapshots_and_cache_changes<br>A doc/CHANGES-staging/chan_pjsip_PJSIP_PARSE_URI<br>A doc/CHANGES-staging/chan_sip_deprecated<br>A doc/CHANGES-staging/channels_remove_stasis_cache_for_channel_snapshots<br>A doc/CHANGES-staging/core_ast_bt_get_symbols_return<br>A doc/CHANGES-staging/features_automon_automixmon<br>A doc/CHANGES-staging/pbx_config_multiple_global_sections<br>A doc/CHANGES-staging/res_pjsip_send_contact_status_on_update_registration<br>A doc/CHANGES-staging/res_pjsip_trust_and_send_connected_line_options<br>A doc/CHANGES-staging/res_rtp_asterisk_strictrtp_seqno_option<br>A doc/UPGRADE-staging/README<br>A doc/UPGRADE-staging/bridge_snapshots_and_cache_changes<br>A doc/UPGRADE-staging/chan_sip_deprecated<br>A doc/UPGRADE-staging/channels_remove_stasis_cache_for_channel_snapshots<br>A doc/UPGRADE-staging/func_callerid_remove_CALLERPRES<br>A doc/UPGRADE-staging/res_parking_PARKINGSLOT_no_longer_set<br>A doc/UPGRADE-staging/res_xmpp_remove_JabberStatus_application<br>21 files changed, 251 insertions(+), 218 deletions(-)<br><br></pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;">git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/45/10945/1</pre><pre style="font-family: monospace,monospace; white-space: pre-wrap;"><span>diff --git a/CHANGES b/CHANGES</span><br><span>index 23612fb..62698a9 100644</span><br><span>--- a/CHANGES</span><br><span>+++ b/CHANGES</span><br><span>@@ -9,133 +9,6 @@</span><br><span> ==============================================================================</span><br><span> </span><br><span> ------------------------------------------------------------------------------</span><br><span>---- Functionality changes from Asterisk 16 to Asterisk 17 --------------------</span><br><span>-------------------------------------------------------------------------------</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-chan_sip</span><br><span>-------------------</span><br><span style="color: hsl(0, 100%, 40%);">- * The chan_sip module is now deprecated, users should migrate to the</span><br><span style="color: hsl(0, 100%, 40%);">- replacement module chan_pjsip. See guides at the Asterisk Wiki:</span><br><span style="color: hsl(0, 100%, 40%);">- https://wiki.asterisk.org/wiki/x/tAHOAQ</span><br><span style="color: hsl(0, 100%, 40%);">- https://wiki.asterisk.org/wiki/x/hYCLAQ</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-Channels</span><br><span>-------------------</span><br><span style="color: hsl(0, 100%, 40%);">- * The core no longer uses the stasis cache for channels snapshots.</span><br><span style="color: hsl(0, 100%, 40%);">- The following APIs are no longer available:</span><br><span style="color: hsl(0, 100%, 40%);">- ast_channel_topic_cached()</span><br><span style="color: hsl(0, 100%, 40%);">- ast_channel_topic_all_cached()</span><br><span style="color: hsl(0, 100%, 40%);">- The ast_channel_cache_all() and ast_channel_cache_by_name() functions</span><br><span style="color: hsl(0, 100%, 40%);">- now returns an ao2_container of ast_channel_snapshots rather than a</span><br><span style="color: hsl(0, 100%, 40%);">- container of stasis_messages therefore you can't call stasis_cache</span><br><span style="color: hsl(0, 100%, 40%);">- functions on it.</span><br><span style="color: hsl(0, 100%, 40%);">- The ast_channel_topic_all() function now returns a normal topic,</span><br><span style="color: hsl(0, 100%, 40%);">- not a cached one so you can't use stasis cache functions on it either.</span><br><span style="color: hsl(0, 100%, 40%);">- The ast_channel_snapshot_type() stasis message now has the</span><br><span style="color: hsl(0, 100%, 40%);">- ast_channel_snapshot_update structure as it's data.</span><br><span style="color: hsl(0, 100%, 40%);">- ast_channel_snapshot_get_latest() still returns the latest snapshot.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-Bridging</span><br><span>-------------------</span><br><span style="color: hsl(0, 100%, 40%);">- * The bridging core no longer uses the stasis cache for bridge</span><br><span style="color: hsl(0, 100%, 40%);">- snapshots. The latest bridge snapshot is now stored on the</span><br><span style="color: hsl(0, 100%, 40%);">- ast_bridge structure itself.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">- * The following APIs are no longer available since the stasis cache</span><br><span style="color: hsl(0, 100%, 40%);">- is no longer used:</span><br><span style="color: hsl(0, 100%, 40%);">- ast_bridge_topic_cached()</span><br><span style="color: hsl(0, 100%, 40%);">- ast_bridge_topic_all_cached()</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">- * A topic pool is now used for individual bridge topics.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">- * The ast_bridge_cache() function was removed since there's no</span><br><span style="color: hsl(0, 100%, 40%);">- longer a separate container of snapshots.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">- * A new function "ast_bridges()" was created to retrieve the</span><br><span style="color: hsl(0, 100%, 40%);">- container of all bridges. Users formerly calling</span><br><span style="color: hsl(0, 100%, 40%);">- ast_bridge_cache() can use the new function to iterate over</span><br><span style="color: hsl(0, 100%, 40%);">- bridges and retrieve the latest snapshot directly from the</span><br><span style="color: hsl(0, 100%, 40%);">- bridge.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">- * The ast_bridge_snapshot_get_latest() function was renamed to</span><br><span style="color: hsl(0, 100%, 40%);">- ast_bridge_get_snapshot_by_uniqueid().</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">- * A new function "ast_bridge_get_snapshot()" was created to retrieve</span><br><span style="color: hsl(0, 100%, 40%);">- the bridge snapshot directly from the bridge structure.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">- * The ast_bridge_topic_all() function now returns a normal topic</span><br><span style="color: hsl(0, 100%, 40%);">- not a cached one so you can't use stasis cache functions on it</span><br><span style="color: hsl(0, 100%, 40%);">- either.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">- * The ast_bridge_snapshot_type() stasis message now has the</span><br><span style="color: hsl(0, 100%, 40%);">- ast_bridge_snapshot_update structure as it's data. It contains</span><br><span style="color: hsl(0, 100%, 40%);">- the last snapshot and the new one.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span>-------------------------------------------------------------------------------</span><br><span>---- Functionality changes from Asterisk 16.1.0 to Asterisk 16.2.0 ------------</span><br><span>-------------------------------------------------------------------------------</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-res_pjsip</span><br><span>-------------------</span><br><span style="color: hsl(0, 100%, 40%);">- * Added "send_contact_status_on_update_registration" global configuration option</span><br><span style="color: hsl(0, 100%, 40%);">- to enable sending AMI ContactStatus event when a device refreshes its registration.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-Features</span><br><span>-------------------</span><br><span style="color: hsl(0, 100%, 40%);">- * Before Asterisk 12, when using the automon or automixmon features defined</span><br><span style="color: hsl(0, 100%, 40%);">- in features.conf, a channel variable (TOUCH_MIXMONITOR_OUTPUT) was set on</span><br><span style="color: hsl(0, 100%, 40%);">- both channels, indicating the filename of the recording.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">- When bridging was overhauled in Asterisk 12, the behavior was changed such</span><br><span style="color: hsl(0, 100%, 40%);">- that the variable was only set on the peer channel and not on the channel</span><br><span style="color: hsl(0, 100%, 40%);">- that initiated the automon or automixmon.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">- The previous behavior has been restored so both channels receive the</span><br><span style="color: hsl(0, 100%, 40%);">- channel variable when one of these features is invoked.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-app_voicemail</span><br><span>-------------------</span><br><span style="color: hsl(0, 100%, 40%);">- * You can now specify a special context with the "aliasescontext" parameter</span><br><span style="color: hsl(0, 100%, 40%);">- in voicemail.conf which will allow you to create aliases for physical</span><br><span style="color: hsl(0, 100%, 40%);">- mailboxes.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span>-------------------------------------------------------------------------------</span><br><span>---- Functionality changes from Asterisk 16.0.0 to Asterisk 16.1.0 ------------</span><br><span>-------------------------------------------------------------------------------</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-pbx_config</span><br><span>-------------------</span><br><span style="color: hsl(0, 100%, 40%);">- * pbx_config will now find and process multiple 'globals' sections from</span><br><span style="color: hsl(0, 100%, 40%);">- extensions.conf. Variables are processed in the order they are found</span><br><span style="color: hsl(0, 100%, 40%);">- and duplicate variables overwrite the previous value.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-chan_pjsip</span><br><span>-------------------</span><br><span style="color: hsl(0, 100%, 40%);">- * New dialplan function PJSIP_PARSE_URI added to parse an URI and return</span><br><span style="color: hsl(0, 100%, 40%);">- a specified part of the URI.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-Core</span><br><span>-------------------</span><br><span style="color: hsl(0, 100%, 40%);">- * ast_bt_get_symbols() now returns a vector of strings instead of an</span><br><span style="color: hsl(0, 100%, 40%);">- array of strings. This must be freed with ast_bt_free_symbols.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-res_pjsip</span><br><span>-------------------</span><br><span style="color: hsl(0, 100%, 40%);">- * New options 'trust_connected_line' and 'send_connected_line' have been</span><br><span style="color: hsl(0, 100%, 40%);">- added to the endpoint. The option 'trust_connected_line' is to control</span><br><span style="color: hsl(0, 100%, 40%);">- if connected line updates are accepted from this endpoint.</span><br><span style="color: hsl(0, 100%, 40%);">- The option 'send_connected_line' is to control if connected line updates</span><br><span style="color: hsl(0, 100%, 40%);">- can be sent to this endpoint.</span><br><span style="color: hsl(0, 100%, 40%);">- The default value is 'yes' for both options.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-res_rtp_asterisk</span><br><span>-------------------</span><br><span style="color: hsl(0, 100%, 40%);">- * The existing strictrtp option in rtp.conf has a new choice availabe, called</span><br><span style="color: hsl(0, 100%, 40%);">- 'seqno', which behaves the same way as setting strictrtp to 'yes', but will</span><br><span style="color: hsl(0, 100%, 40%);">- ignore the time interval during learning so that bursts of packets can still</span><br><span style="color: hsl(0, 100%, 40%);">- trigger learning our source.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span>-------------------------------------------------------------------------------</span><br><span> --- Functionality changes from Asterisk 15 to Asterisk 16 --------------------</span><br><span> ------------------------------------------------------------------------------</span><br><span> </span><br><span>diff --git a/UPGRADE.txt b/UPGRADE.txt</span><br><span>deleted file mode 100644</span><br><span>index 53e9c4a..0000000</span><br><span>--- a/UPGRADE.txt</span><br><span>+++ /dev/null</span><br><span>@@ -1,91 +0,0 @@</span><br><span style="color: hsl(0, 100%, 40%);">-===========================================================</span><br><span style="color: hsl(0, 100%, 40%);">-===</span><br><span style="color: hsl(0, 100%, 40%);">-=== Information for upgrading between Asterisk versions</span><br><span style="color: hsl(0, 100%, 40%);">-===</span><br><span style="color: hsl(0, 100%, 40%);">-=== These files document all the changes that MUST be taken</span><br><span style="color: hsl(0, 100%, 40%);">-=== into account when upgrading between the Asterisk</span><br><span style="color: hsl(0, 100%, 40%);">-=== versions listed below. These changes may require that</span><br><span style="color: hsl(0, 100%, 40%);">-=== you modify your configuration files, dialplan or (in</span><br><span style="color: hsl(0, 100%, 40%);">-=== some cases) source code if you have your own Asterisk</span><br><span style="color: hsl(0, 100%, 40%);">-=== modules or patches. These files also include advance</span><br><span style="color: hsl(0, 100%, 40%);">-=== notice of any functionality that has been marked as</span><br><span style="color: hsl(0, 100%, 40%);">-=== 'deprecated' and may be removed in a future release,</span><br><span style="color: hsl(0, 100%, 40%);">-=== along with the suggested replacement functionality.</span><br><span style="color: hsl(0, 100%, 40%);">-===</span><br><span style="color: hsl(0, 100%, 40%);">-=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2</span><br><span style="color: hsl(0, 100%, 40%);">-=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4</span><br><span style="color: hsl(0, 100%, 40%);">-=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6</span><br><span style="color: hsl(0, 100%, 40%);">-=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8</span><br><span style="color: hsl(0, 100%, 40%);">-=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10</span><br><span style="color: hsl(0, 100%, 40%);">-=== UPGRADE-11.txt -- Upgrade info for 10 to 11</span><br><span style="color: hsl(0, 100%, 40%);">-=== UPGRADE-12.txt -- Upgrade info for 11 to 12</span><br><span style="color: hsl(0, 100%, 40%);">-=== UPGRADE-13.txt -- Upgrade info for 12 to 13</span><br><span style="color: hsl(0, 100%, 40%);">-=== UPGRADE-14.txt -- Upgrade info for 13 to 14</span><br><span style="color: hsl(0, 100%, 40%);">-=== UPGRADE-15.txt -- Upgrade info for 14 to 15</span><br><span style="color: hsl(0, 100%, 40%);">-=== UPGRADE-16.txt -- Upgrade info for 15 to 16</span><br><span style="color: hsl(0, 100%, 40%);">-===========================================================</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-New in 17.0.0:</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-chan_sip:</span><br><span style="color: hsl(0, 100%, 40%);">- - The chan_sip module is now deprecated, users should migrate to the</span><br><span style="color: hsl(0, 100%, 40%);">- replacement module chan_pjsip. See guides at the Asterisk Wiki:</span><br><span style="color: hsl(0, 100%, 40%);">- https://wiki.asterisk.org/wiki/x/tAHOAQ</span><br><span style="color: hsl(0, 100%, 40%);">- https://wiki.asterisk.org/wiki/x/hYCLAQ</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-func_callerid:</span><br><span style="color: hsl(0, 100%, 40%);">- - The CALLERPRES() dialplan function, deprecated in Asterisk 1.8, has been</span><br><span style="color: hsl(0, 100%, 40%);">- removed.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-res_parking:</span><br><span style="color: hsl(0, 100%, 40%);">- - The PARKINGSLOT channel variable, deprecated in Asterisk 12 in favor of the</span><br><span style="color: hsl(0, 100%, 40%);">- PARKING_SPACE channel variable, will no longer be set.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-res_xmpp:</span><br><span style="color: hsl(0, 100%, 40%);">- - The JabberStatus application, deprecated in Asterisk 12, has been removed.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-Channels:</span><br><span style="color: hsl(0, 100%, 40%);">- - The core no longer uses the stasis cache for channels snapshots.</span><br><span style="color: hsl(0, 100%, 40%);">- The following APIs are no longer available:</span><br><span style="color: hsl(0, 100%, 40%);">- ast_channel_topic_cached()</span><br><span style="color: hsl(0, 100%, 40%);">- ast_channel_topic_all_cached()</span><br><span style="color: hsl(0, 100%, 40%);">- The ast_channel_cache_all() and ast_channel_cache_by_name() functions</span><br><span style="color: hsl(0, 100%, 40%);">- now returns an ao2_container of ast_channel_snapshots rather than a</span><br><span style="color: hsl(0, 100%, 40%);">- container of stasis_messages therefore you can't call stasis_cache</span><br><span style="color: hsl(0, 100%, 40%);">- functions on it.</span><br><span style="color: hsl(0, 100%, 40%);">- The ast_channel_topic_all() function now returns a normal topic,</span><br><span style="color: hsl(0, 100%, 40%);">- not a cached one so you can't use stasis cache functions on it either.</span><br><span style="color: hsl(0, 100%, 40%);">- The ast_channel_snapshot_type() stasis message now has the</span><br><span style="color: hsl(0, 100%, 40%);">- ast_channel_snapshot_update structure as it's data.</span><br><span style="color: hsl(0, 100%, 40%);">- ast_channel_snapshot_get_latest() still returns the latest snapshot.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-Applications</span><br><span style="color: hsl(0, 100%, 40%);">- - The JabberStatus application, deprecated in Asterisk 12, has been removed.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span style="color: hsl(0, 100%, 40%);">-Bridging</span><br><span style="color: hsl(0, 100%, 40%);">- - The bridging core no longer uses the stasis cache for bridge</span><br><span style="color: hsl(0, 100%, 40%);">- snapshots. The latest bridge snapshot is now stored on the</span><br><span style="color: hsl(0, 100%, 40%);">- ast_bridge structure itself.</span><br><span style="color: hsl(0, 100%, 40%);">- - The following APIs are no longer available since the stasis cache</span><br><span style="color: hsl(0, 100%, 40%);">- is no longer used:</span><br><span style="color: hsl(0, 100%, 40%);">- ast_bridge_topic_cached()</span><br><span style="color: hsl(0, 100%, 40%);">- ast_bridge_topic_all_cached()</span><br><span style="color: hsl(0, 100%, 40%);">- - A topic pool is now used for individual bridge topics.</span><br><span style="color: hsl(0, 100%, 40%);">- - The ast_bridge_cache() function was removed since there's no</span><br><span style="color: hsl(0, 100%, 40%);">- longer a separate container of snapshots.</span><br><span style="color: hsl(0, 100%, 40%);">- - A new function "ast_bridges()" was created to retrieve the</span><br><span style="color: hsl(0, 100%, 40%);">- container of all bridges. Users formerly calling</span><br><span style="color: hsl(0, 100%, 40%);">- ast_bridge_cache() can use the new function to iterate over</span><br><span style="color: hsl(0, 100%, 40%);">- bridges and retrieve the latest snapshot directly from the</span><br><span style="color: hsl(0, 100%, 40%);">- bridge.</span><br><span style="color: hsl(0, 100%, 40%);">- - The ast_bridge_snapshot_get_latest() function was renamed to</span><br><span style="color: hsl(0, 100%, 40%);">- ast_bridge_get_snapshot_by_uniqueid().</span><br><span style="color: hsl(0, 100%, 40%);">- - A new function "ast_bridge_get_snapshot()" was created to retrieve</span><br><span style="color: hsl(0, 100%, 40%);">- the bridge snapshot directly from the bridge structure.</span><br><span style="color: hsl(0, 100%, 40%);">- - The ast_bridge_topic_all() function now returns a normal topic</span><br><span style="color: hsl(0, 100%, 40%);">- not a cached one so you can't use stasis cache functions on it</span><br><span style="color: hsl(0, 100%, 40%);">- either.</span><br><span style="color: hsl(0, 100%, 40%);">- - The ast_bridge_snapshot_type() stasis message now has the</span><br><span style="color: hsl(0, 100%, 40%);">- ast_bridge_snapshot_update structure as it's data. It contains</span><br><span style="color: hsl(0, 100%, 40%);">- the last snapshot and the new one.</span><br><span style="color: hsl(0, 100%, 40%);">-</span><br><span>diff --git a/doc/CHANGES-staging/README b/doc/CHANGES-staging/README</span><br><span>new file mode 100644</span><br><span>index 0000000..9c3c79f</span><br><span>--- /dev/null</span><br><span>+++ b/doc/CHANGES-staging/README</span><br><span>@@ -0,0 +1,22 @@</span><br><span style="color: hsl(120, 100%, 40%);">+DO NOT REMOVE THIS FILE!</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The only files that should be added to this directory are ones that will be</span><br><span style="color: hsl(120, 100%, 40%);">+used by the release script to update the CHANGES file automatically. The only</span><br><span style="color: hsl(120, 100%, 40%);">+time that it is necessary to add something to the CHANGES-staging directory is</span><br><span style="color: hsl(120, 100%, 40%);">+if you are either adding a new feature to Asterisk or adding new functionality</span><br><span style="color: hsl(120, 100%, 40%);">+to an existing feature. The file does not need to have a meaningful name, but</span><br><span style="color: hsl(120, 100%, 40%);">+it probably should. If there are multiple items that need documenting, each can</span><br><span style="color: hsl(120, 100%, 40%);">+be separated with a subject line, which should always start with "SUBJECT:",</span><br><span style="color: hsl(120, 100%, 40%);">+followed by the subject of the change. For example, if you are making a change</span><br><span style="color: hsl(120, 100%, 40%);">+to PJSIP, then you might add the file "res_pjsip_my_cool_feature" to this</span><br><span style="color: hsl(120, 100%, 40%);">+directory, with a short description of what it does. Here's an example:</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+SUBJECT: res_pjsip</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Here's a pretty good description of my new feature that explains exactly what</span><br><span style="color: hsl(120, 100%, 40%);">+it does and how to use it.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+SUBJECT: core</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Here's another description of something else I added that is a big enough</span><br><span style="color: hsl(120, 100%, 40%);">+change to warrant another entry in the CHANGES file.</span><br><span>diff --git a/doc/CHANGES-staging/app_voicemail_aliasescontext b/doc/CHANGES-staging/app_voicemail_aliasescontext</span><br><span>new file mode 100644</span><br><span>index 0000000..e56dabd</span><br><span>--- /dev/null</span><br><span>+++ b/doc/CHANGES-staging/app_voicemail_aliasescontext</span><br><span>@@ -0,0 +1,5 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: app_voicemail</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+You can now specify a special context with the "aliasescontext" parameter</span><br><span style="color: hsl(120, 100%, 40%);">+in voicemail.conf which will allow you to create aliases for physical</span><br><span style="color: hsl(120, 100%, 40%);">+mailboxes.</span><br><span>diff --git a/doc/CHANGES-staging/bridge_snapshots_and_cache_changes b/doc/CHANGES-staging/bridge_snapshots_and_cache_changes</span><br><span>new file mode 100644</span><br><span>index 0000000..ebcca2e</span><br><span>--- /dev/null</span><br><span>+++ b/doc/CHANGES-staging/bridge_snapshots_and_cache_changes</span><br><span>@@ -0,0 +1,51 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The bridging core no longer uses the stasis cache for bridge</span><br><span style="color: hsl(120, 100%, 40%);">+snapshots. The latest bridge snapshot is now stored on the</span><br><span style="color: hsl(120, 100%, 40%);">+ast_bridge structure itself.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The following APIs are no longer available since the stasis cache</span><br><span style="color: hsl(120, 100%, 40%);">+is no longer used:</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_bridge_topic_cached()</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_bridge_topic_all_cached()</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+A topic pool is now used for individual bridge topics.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The ast_bridge_cache() function was removed since there's no</span><br><span style="color: hsl(120, 100%, 40%);">+longer a separate container of snapshots.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+A new function "ast_bridges()" was created to retrieve the</span><br><span style="color: hsl(120, 100%, 40%);">+container of all bridges. Users formerly calling</span><br><span style="color: hsl(120, 100%, 40%);">+ast_bridge_cache() can use the new function to iterate over</span><br><span style="color: hsl(120, 100%, 40%);">+bridges and retrieve the latest snapshot directly from the</span><br><span style="color: hsl(120, 100%, 40%);">+bridge.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The ast_bridge_snapshot_get_latest() function was renamed to</span><br><span style="color: hsl(120, 100%, 40%);">+ast_bridge_get_snapshot_by_uniqueid().</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+A new function "ast_bridge_get_snapshot()" was created to retrieve</span><br><span style="color: hsl(120, 100%, 40%);">+the bridge snapshot directly from the bridge structure.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The ast_bridge_topic_all() function now returns a normal topic</span><br><span style="color: hsl(120, 100%, 40%);">+not a cached one so you can't use stasis cache functions on it</span><br><span style="color: hsl(120, 100%, 40%);">+either.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The ast_bridge_snapshot_type() stasis message now has the</span><br><span style="color: hsl(120, 100%, 40%);">+ast_bridge_snapshot_update structure as it's data. It contains</span><br><span style="color: hsl(120, 100%, 40%);">+the last snapshot and the new one.</span><br><span>diff --git a/doc/CHANGES-staging/chan_pjsip_PJSIP_PARSE_URI b/doc/CHANGES-staging/chan_pjsip_PJSIP_PARSE_URI</span><br><span>new file mode 100644</span><br><span>index 0000000..85854ee</span><br><span>--- /dev/null</span><br><span>+++ b/doc/CHANGES-staging/chan_pjsip_PJSIP_PARSE_URI</span><br><span>@@ -0,0 +1,4 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: chan_pjsip</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+New dialplan function PJSIP_PARSE_URI added to parse an URI and return</span><br><span style="color: hsl(120, 100%, 40%);">+a specified part of the URI.</span><br><span>diff --git a/doc/CHANGES-staging/chan_sip_deprecated b/doc/CHANGES-staging/chan_sip_deprecated</span><br><span>new file mode 100644</span><br><span>index 0000000..04ac21c</span><br><span>--- /dev/null</span><br><span>+++ b/doc/CHANGES-staging/chan_sip_deprecated</span><br><span>@@ -0,0 +1,6 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: chan_sip</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The chan_sip module is now deprecated, users should migrate to the</span><br><span style="color: hsl(120, 100%, 40%);">+replacement module chan_pjsip. See guides at the Asterisk Wiki:</span><br><span style="color: hsl(120, 100%, 40%);">+ https://wiki.asterisk.org/wiki/x/tAHOAQ</span><br><span style="color: hsl(120, 100%, 40%);">+ https://wiki.asterisk.org/wiki/x/hYCLAQ</span><br><span>diff --git a/doc/CHANGES-staging/channels_remove_stasis_cache_for_channel_snapshots b/doc/CHANGES-staging/channels_remove_stasis_cache_for_channel_snapshots</span><br><span>new file mode 100644</span><br><span>index 0000000..1743fea</span><br><span>--- /dev/null</span><br><span>+++ b/doc/CHANGES-staging/channels_remove_stasis_cache_for_channel_snapshots</span><br><span>@@ -0,0 +1,15 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Channels</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The core no longer uses the stasis cache for channels snapshots.</span><br><span style="color: hsl(120, 100%, 40%);">+The following APIs are no longer available:</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_channel_topic_cached()</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_channel_topic_all_cached()</span><br><span style="color: hsl(120, 100%, 40%);">+The ast_channel_cache_all() and ast_channel_cache_by_name() functions</span><br><span style="color: hsl(120, 100%, 40%);">+now returns an ao2_container of ast_channel_snapshots rather than a</span><br><span style="color: hsl(120, 100%, 40%);">+container of stasis_messages therefore you can't call stasis_cache</span><br><span style="color: hsl(120, 100%, 40%);">+functions on it.</span><br><span style="color: hsl(120, 100%, 40%);">+The ast_channel_topic_all() function now returns a normal topic,</span><br><span style="color: hsl(120, 100%, 40%);">+not a cached one so you can't use stasis cache functions on it either.</span><br><span style="color: hsl(120, 100%, 40%);">+The ast_channel_snapshot_type() stasis message now has the</span><br><span style="color: hsl(120, 100%, 40%);">+ast_channel_snapshot_update structure as it's data.</span><br><span style="color: hsl(120, 100%, 40%);">+ast_channel_snapshot_get_latest() still returns the latest snapshot.</span><br><span>diff --git a/doc/CHANGES-staging/core_ast_bt_get_symbols_return b/doc/CHANGES-staging/core_ast_bt_get_symbols_return</span><br><span>new file mode 100644</span><br><span>index 0000000..a482a4c</span><br><span>--- /dev/null</span><br><span>+++ b/doc/CHANGES-staging/core_ast_bt_get_symbols_return</span><br><span>@@ -0,0 +1,4 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Core</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+ast_bt_get_symbols() now returns a vector of strings instead of an</span><br><span style="color: hsl(120, 100%, 40%);">+array of strings. This must be freed with ast_bt_free_symbols.</span><br><span>diff --git a/doc/CHANGES-staging/features_automon_automixmon b/doc/CHANGES-staging/features_automon_automixmon</span><br><span>new file mode 100644</span><br><span>index 0000000..b97c883</span><br><span>--- /dev/null</span><br><span>+++ b/doc/CHANGES-staging/features_automon_automixmon</span><br><span>@@ -0,0 +1,12 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Features</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Before Asterisk 12, when using the automon or automixmon features defined</span><br><span style="color: hsl(120, 100%, 40%);">+in features.conf, a channel variable (TOUCH_MIXMONITOR_OUTPUT) was set on</span><br><span style="color: hsl(120, 100%, 40%);">+both channels, indicating the filename of the recording.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+When bridging was overhauled in Asterisk 12, the behavior was changed such</span><br><span style="color: hsl(120, 100%, 40%);">+that the variable was only set on the peer channel and not on the channel</span><br><span style="color: hsl(120, 100%, 40%);">+that initiated the automon or automixmon.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The previous behavior has been restored so both channels receive the</span><br><span style="color: hsl(120, 100%, 40%);">+channel variable when one of these features is invoked.</span><br><span>diff --git a/doc/CHANGES-staging/pbx_config_multiple_global_sections b/doc/CHANGES-staging/pbx_config_multiple_global_sections</span><br><span>new file mode 100644</span><br><span>index 0000000..822cf42</span><br><span>--- /dev/null</span><br><span>+++ b/doc/CHANGES-staging/pbx_config_multiple_global_sections</span><br><span>@@ -0,0 +1,5 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: pbx_config</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+pbx_config will now find and process multiple 'globals' sections from</span><br><span style="color: hsl(120, 100%, 40%);">+extensions.conf. Variables are processed in the order they are found</span><br><span style="color: hsl(120, 100%, 40%);">+and duplicate variables overwrite the previous value.</span><br><span>diff --git a/doc/CHANGES-staging/res_pjsip_send_contact_status_on_update_registration b/doc/CHANGES-staging/res_pjsip_send_contact_status_on_update_registration</span><br><span>new file mode 100644</span><br><span>index 0000000..e27a0c0</span><br><span>--- /dev/null</span><br><span>+++ b/doc/CHANGES-staging/res_pjsip_send_contact_status_on_update_registration</span><br><span>@@ -0,0 +1,4 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: res_pjsip</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Added "send_contact_status_on_update_registration" global configuration option</span><br><span style="color: hsl(120, 100%, 40%);">+to enable sending AMI ContactStatus event when a device refreshes its registration.</span><br><span>diff --git a/doc/CHANGES-staging/res_pjsip_trust_and_send_connected_line_options b/doc/CHANGES-staging/res_pjsip_trust_and_send_connected_line_options</span><br><span>new file mode 100644</span><br><span>index 0000000..44a4c05</span><br><span>--- /dev/null</span><br><span>+++ b/doc/CHANGES-staging/res_pjsip_trust_and_send_connected_line_options</span><br><span>@@ -0,0 +1,8 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: res_pjsip</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+New options 'trust_connected_line' and 'send_connected_line' have been</span><br><span style="color: hsl(120, 100%, 40%);">+added to the endpoint. The option 'trust_connected_line' is to control</span><br><span style="color: hsl(120, 100%, 40%);">+if connected line updates are accepted from this endpoint.</span><br><span style="color: hsl(120, 100%, 40%);">+The option 'send_connected_line' is to control if connected line updates</span><br><span style="color: hsl(120, 100%, 40%);">+can be sent to this endpoint.</span><br><span style="color: hsl(120, 100%, 40%);">+The default value is 'yes' for both options.</span><br><span>diff --git a/doc/CHANGES-staging/res_rtp_asterisk_strictrtp_seqno_option b/doc/CHANGES-staging/res_rtp_asterisk_strictrtp_seqno_option</span><br><span>new file mode 100644</span><br><span>index 0000000..a2db17b</span><br><span>--- /dev/null</span><br><span>+++ b/doc/CHANGES-staging/res_rtp_asterisk_strictrtp_seqno_option</span><br><span>@@ -0,0 +1,6 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: res_rtp_asterisk</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The existing strictrtp option in rtp.conf has a new choice availabe, called</span><br><span style="color: hsl(120, 100%, 40%);">+'seqno', which behaves the same way as setting strictrtp to 'yes', but will</span><br><span style="color: hsl(120, 100%, 40%);">+ignore the time interval during learning so that bursts of packets can still</span><br><span style="color: hsl(120, 100%, 40%);">+trigger learning our source.</span><br><span>diff --git a/doc/UPGRADE-staging/README b/doc/UPGRADE-staging/README</span><br><span>new file mode 100644</span><br><span>index 0000000..95ec1ff</span><br><span>--- /dev/null</span><br><span>+++ b/doc/UPGRADE-staging/README</span><br><span>@@ -0,0 +1,22 @@</span><br><span style="color: hsl(120, 100%, 40%);">+DO NOT REMOVE THIS FILE!</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The only files that should be added to this directory are ones that will be</span><br><span style="color: hsl(120, 100%, 40%);">+used by the release script to update the UPGRADE.txt file automatically. The</span><br><span style="color: hsl(120, 100%, 40%);">+only time that it is necessary to add something to the UPGRADE-staging directory</span><br><span style="color: hsl(120, 100%, 40%);">+is if you are making a breaking change to an existing feature in Asterisk. The</span><br><span style="color: hsl(120, 100%, 40%);">+file does not need to have a meaningful name, but it probably should. If there</span><br><span style="color: hsl(120, 100%, 40%);">+are multiple items that need documenting, each can be separated with a subject</span><br><span style="color: hsl(120, 100%, 40%);">+line, which should always start with "SUBJECT:", followed by the subject of the</span><br><span style="color: hsl(120, 100%, 40%);">+change. For example, if you are making a change to PJSIP, then you might add the</span><br><span style="color: hsl(120, 100%, 40%);">+file "res_pjsip_breaking_change" to this directory, with a short description of</span><br><span style="color: hsl(120, 100%, 40%);">+what it does. Here's an example:</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+SUBJECT: res_pjsip</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Here's a pretty good description of what I changed that explains exactly what</span><br><span style="color: hsl(120, 100%, 40%);">+it does and why it breaks things (and why they needed to be broken).</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+SUBJECT: core</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Here's another description of something else I added that is a big enough</span><br><span style="color: hsl(120, 100%, 40%);">+change to warrant another entry in the UPDATE.txt file.</span><br><span>diff --git a/doc/UPGRADE-staging/bridge_snapshots_and_cache_changes b/doc/UPGRADE-staging/bridge_snapshots_and_cache_changes</span><br><span>new file mode 100644</span><br><span>index 0000000..ebcca2e</span><br><span>--- /dev/null</span><br><span>+++ b/doc/UPGRADE-staging/bridge_snapshots_and_cache_changes</span><br><span>@@ -0,0 +1,51 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The bridging core no longer uses the stasis cache for bridge</span><br><span style="color: hsl(120, 100%, 40%);">+snapshots. The latest bridge snapshot is now stored on the</span><br><span style="color: hsl(120, 100%, 40%);">+ast_bridge structure itself.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The following APIs are no longer available since the stasis cache</span><br><span style="color: hsl(120, 100%, 40%);">+is no longer used:</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_bridge_topic_cached()</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_bridge_topic_all_cached()</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+A topic pool is now used for individual bridge topics.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The ast_bridge_cache() function was removed since there's no</span><br><span style="color: hsl(120, 100%, 40%);">+longer a separate container of snapshots.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+A new function "ast_bridges()" was created to retrieve the</span><br><span style="color: hsl(120, 100%, 40%);">+container of all bridges. Users formerly calling</span><br><span style="color: hsl(120, 100%, 40%);">+ast_bridge_cache() can use the new function to iterate over</span><br><span style="color: hsl(120, 100%, 40%);">+bridges and retrieve the latest snapshot directly from the</span><br><span style="color: hsl(120, 100%, 40%);">+bridge.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The ast_bridge_snapshot_get_latest() function was renamed to</span><br><span style="color: hsl(120, 100%, 40%);">+ast_bridge_get_snapshot_by_uniqueid().</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+A new function "ast_bridge_get_snapshot()" was created to retrieve</span><br><span style="color: hsl(120, 100%, 40%);">+the bridge snapshot directly from the bridge structure.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The ast_bridge_topic_all() function now returns a normal topic</span><br><span style="color: hsl(120, 100%, 40%);">+not a cached one so you can't use stasis cache functions on it</span><br><span style="color: hsl(120, 100%, 40%);">+either.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Bridging</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The ast_bridge_snapshot_type() stasis message now has the</span><br><span style="color: hsl(120, 100%, 40%);">+ast_bridge_snapshot_update structure as it's data. It contains</span><br><span style="color: hsl(120, 100%, 40%);">+the last snapshot and the new one.</span><br><span>diff --git a/doc/UPGRADE-staging/chan_sip_deprecated b/doc/UPGRADE-staging/chan_sip_deprecated</span><br><span>new file mode 100644</span><br><span>index 0000000..04ac21c</span><br><span>--- /dev/null</span><br><span>+++ b/doc/UPGRADE-staging/chan_sip_deprecated</span><br><span>@@ -0,0 +1,6 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: chan_sip</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The chan_sip module is now deprecated, users should migrate to the</span><br><span style="color: hsl(120, 100%, 40%);">+replacement module chan_pjsip. See guides at the Asterisk Wiki:</span><br><span style="color: hsl(120, 100%, 40%);">+ https://wiki.asterisk.org/wiki/x/tAHOAQ</span><br><span style="color: hsl(120, 100%, 40%);">+ https://wiki.asterisk.org/wiki/x/hYCLAQ</span><br><span>diff --git a/doc/UPGRADE-staging/channels_remove_stasis_cache_for_channel_snapshots b/doc/UPGRADE-staging/channels_remove_stasis_cache_for_channel_snapshots</span><br><span>new file mode 100644</span><br><span>index 0000000..1743fea</span><br><span>--- /dev/null</span><br><span>+++ b/doc/UPGRADE-staging/channels_remove_stasis_cache_for_channel_snapshots</span><br><span>@@ -0,0 +1,15 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Channels</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The core no longer uses the stasis cache for channels snapshots.</span><br><span style="color: hsl(120, 100%, 40%);">+The following APIs are no longer available:</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_channel_topic_cached()</span><br><span style="color: hsl(120, 100%, 40%);">+ ast_channel_topic_all_cached()</span><br><span style="color: hsl(120, 100%, 40%);">+The ast_channel_cache_all() and ast_channel_cache_by_name() functions</span><br><span style="color: hsl(120, 100%, 40%);">+now returns an ao2_container of ast_channel_snapshots rather than a</span><br><span style="color: hsl(120, 100%, 40%);">+container of stasis_messages therefore you can't call stasis_cache</span><br><span style="color: hsl(120, 100%, 40%);">+functions on it.</span><br><span style="color: hsl(120, 100%, 40%);">+The ast_channel_topic_all() function now returns a normal topic,</span><br><span style="color: hsl(120, 100%, 40%);">+not a cached one so you can't use stasis cache functions on it either.</span><br><span style="color: hsl(120, 100%, 40%);">+The ast_channel_snapshot_type() stasis message now has the</span><br><span style="color: hsl(120, 100%, 40%);">+ast_channel_snapshot_update structure as it's data.</span><br><span style="color: hsl(120, 100%, 40%);">+ast_channel_snapshot_get_latest() still returns the latest snapshot.</span><br><span>diff --git a/doc/UPGRADE-staging/func_callerid_remove_CALLERPRES b/doc/UPGRADE-staging/func_callerid_remove_CALLERPRES</span><br><span>new file mode 100644</span><br><span>index 0000000..b128cc6</span><br><span>--- /dev/null</span><br><span>+++ b/doc/UPGRADE-staging/func_callerid_remove_CALLERPRES</span><br><span>@@ -0,0 +1,4 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: func_callerid</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The CALLERPRES() dialplan function, deprecated in Asterisk 1.8, has been</span><br><span style="color: hsl(120, 100%, 40%);">+removed.</span><br><span>diff --git a/doc/UPGRADE-staging/res_parking_PARKINGSLOT_no_longer_set b/doc/UPGRADE-staging/res_parking_PARKINGSLOT_no_longer_set</span><br><span>new file mode 100644</span><br><span>index 0000000..1ddcaeb</span><br><span>--- /dev/null</span><br><span>+++ b/doc/UPGRADE-staging/res_parking_PARKINGSLOT_no_longer_set</span><br><span>@@ -0,0 +1,4 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: res_parking</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The PARKINGSLOT channel variable, deprecated in Asterisk 12 in favor of the</span><br><span style="color: hsl(120, 100%, 40%);">+PARKING_SPACE channel variable, will no longer be set.</span><br><span>diff --git a/doc/UPGRADE-staging/res_xmpp_remove_JabberStatus_application b/doc/UPGRADE-staging/res_xmpp_remove_JabberStatus_application</span><br><span>new file mode 100644</span><br><span>index 0000000..b0aac75</span><br><span>--- /dev/null</span><br><span>+++ b/doc/UPGRADE-staging/res_xmpp_remove_JabberStatus_application</span><br><span>@@ -0,0 +1,7 @@</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: res_xmpp</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The JabberStatus application, deprecated in Asterisk 12, has been removed.</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+Subject: Applications</span><br><span style="color: hsl(120, 100%, 40%);">+</span><br><span style="color: hsl(120, 100%, 40%);">+The JabberStatus application, deprecated in Asterisk 12, has been removed.</span><br><span></span><br></pre><p>To view, visit <a href="https://gerrit.asterisk.org/10945">change 10945</a>. To unsubscribe, or for help writing mail filters, visit <a href="https://gerrit.asterisk.org/settings">settings</a>.</p><div itemscope itemtype="http://schema.org/EmailMessage"><div itemscope itemprop="action" itemtype="http://schema.org/ViewAction"><link itemprop="url" href="https://gerrit.asterisk.org/10945"/><meta itemprop="name" content="View Change"/></div></div>
<div style="display:none"> Gerrit-Project: asterisk </div>
<div style="display:none"> Gerrit-Branch: master </div>
<div style="display:none"> Gerrit-MessageType: newchange </div>
<div style="display:none"> Gerrit-Change-Id: I19cf4b569321c88155a65e9b0b80f6d58075dd47 </div>
<div style="display:none"> Gerrit-Change-Number: 10945 </div>
<div style="display:none"> Gerrit-PatchSet: 1 </div>
<div style="display:none"> Gerrit-Owner: Benjamin Keith Ford <bford@digium.com> </div>