Update for 19.4.1

As part of PJSIP 2.11 a behavior change was done to require
a matching remote hostname on an established transport for
secure transports. Since the Websocket transport is considered
a secure transport this caused the existing connection to not
be found and used.

We now set the remote hostname and the transport can be found.

ASTERISK-30065

Change-Id: Ia1cdef33e1411f927985b4b852c95e163c080e94
(cherry picked from commit dc1e446ebd)

.lastclean

@@ -0,0 +1 @@
+40

.version

@@ -0,0 +1 @@
+19.4.1

CHANGES

@@ -12,6 +12,70 @@
 ===
 ==============================================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 19.3.3 to Asterisk 19.4.0 ------------
+------------------------------------------------------------------------------
+
+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.
+
+app_queue
+------------------
+ * The m option now allows an override music on hold
+   class to be specified for the Queue application
+   within the dialplan.
+
+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.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 19.3.1 to Asterisk 19.3.2 ------------
 ------------------------------------------------------------------------------

UPGRADE.txt

@@ -18,6 +18,18 @@
 ===
 ===========================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 19.3.3 to Asterisk 19.4.0 ------------
+------------------------------------------------------------------------------
+
+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 ------------
 ------------------------------------------------------------------------------

asterisk-19.4.1-summary.html

@@ -0,0 +1,13 @@
+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-19.4.1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-19.4.1</h3><h3 align="center">Date: 2022-05-19</h3><h3 align="center">&lt;asteriskteam@digium.com&gt;</h3><hr><h2 align="center">Table of Contents</h2><ol>
+<li><a href="#summary">Summary</a></li>
+<li><a href="#contributors">Contributors</a></li>
+<li><a href="#closed_issues">Closed Issues</a></li>
+<li><a href="#diffstat">Diffstat</a></li>
+</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-19.4.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
+<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
+<tr valign="top"><td width="33%">1 Joshua C. Colp <jcolp@sangoma.com><br/></td><td width="33%"><td width="33%">1 LA <learbia@gmail.com><br/></td></tr>
+</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>Bug</h3><h4>Category: Resources/res_pjsip_transport_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30065">ASTERISK-30065</a>: pjsip: Open Websocket connection is not reused for outgoing requests<br/>Reported by: LA<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3d441e1080d98e6ef23cd9206409eaf84184f153">[3d441e1080]</a> Joshua C. Colp -- res_pjsip_transport_websocket: Also set the remote name.</li>
+</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-30065">ASTERISK-30065</a>: pjsip: Open Websocket connection is not reused for outgoing requests<br/>Reported by: LA<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3d441e1080d98e6ef23cd9206409eaf84184f153">[3d441e1080]</a> Joshua C. Colp -- res_pjsip_transport_websocket: Also set the remote name.</li>
+</ul><br><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>0 files changed</pre><br></html>

asterisk-19.4.1-summary.txt

@@ -0,0 +1,90 @@
+                                Release Summary
+
+                                asterisk-19.4.1
+
+                                Date: 2022-05-19
+
+                           <asteriskteam@digium.com>
+
+     ----------------------------------------------------------------------
+
+                               Table of Contents
+
+    1. Summary
+    2. Contributors
+    3. Closed Issues
+    4. Diffstat
+
+     ----------------------------------------------------------------------
+
+                                    Summary
+
+                                 [Back to Top]
+
+   This release is a point release of an existing major version. The changes
+   included were made to address problems that have been identified in this
+   release series, or are minor, backwards compatible new features or
+   improvements. Users should be able to safely upgrade to this version if
+   this release series is already in use. Users considering upgrading from a
+   previous version are strongly encouraged to review the UPGRADE.txt
+   document as well as the CHANGES document for information about upgrading
+   to this release series.
+
+   The data in this summary reflects changes that have been made since the
+   previous release, asterisk-19.4.0.
+
+     ----------------------------------------------------------------------
+
+                                  Contributors
+
+                                 [Back to Top]
+
+   This table lists the people who have submitted code, those that have
+   tested patches, as well as those that reported issues on the issue tracker
+   that were resolved in this release. For coders, the number is how many of
+   their patches (of any size) were committed into this release. For testers,
+   the number is the number of times their name was listed as assisting with
+   testing a patch. Finally, for reporters, the number is the number of
+   issues that they reported that were affected by commits that went into
+   this release.
+
+   Coders                   Testers                  Reporters                
+   1 Joshua C. Colp                                  1 LA                     
+
+     ----------------------------------------------------------------------
+
+                                 Closed Issues
+
+                                 [Back to Top]
+
+   This is a list of all issues from the issue tracker that were closed by
+   changes that went into this release.
+
+  Bug
+
+    Category: Resources/res_pjsip_transport_websocket
+
+   ASTERISK-30065: pjsip: Open Websocket connection is not reused for
+   outgoing requests
+   Reported by: LA
+     * [3d441e1080] Joshua C. Colp -- res_pjsip_transport_websocket: Also set
+       the remote name.
+
+    Category: pjproject/pjsip
+
+   ASTERISK-30065: pjsip: Open Websocket connection is not reused for
+   outgoing requests
+   Reported by: LA
+     * [3d441e1080] Joshua C. Colp -- res_pjsip_transport_websocket: Also set
+       the remote name.
+
+     ----------------------------------------------------------------------
+
+                                Diffstat Results
+
+                                 [Back to Top]
+
+   This is a summary of the changes to the source code that went into this
+   release that was generated using the diffstat utility.
+
+ 0 files changed

contrib/realtime/mysql/mysql_cdr.sql

@@ -0,0 +1,41 @@
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> 210693f3123d
+
+CREATE TABLE cdr (
+    accountcode VARCHAR(20), 
+    src VARCHAR(80), 
+    dst VARCHAR(80), 
+    dcontext VARCHAR(80), 
+    clid VARCHAR(80), 
+    channel VARCHAR(80), 
+    dstchannel VARCHAR(80), 
+    lastapp VARCHAR(80), 
+    lastdata VARCHAR(80), 
+    start DATETIME, 
+    answer DATETIME, 
+    end DATETIME, 
+    duration INTEGER, 
+    billsec INTEGER, 
+    disposition VARCHAR(45), 
+    amaflags VARCHAR(45), 
+    userfield VARCHAR(256), 
+    uniqueid VARCHAR(150), 
+    linkedid VARCHAR(150), 
+    peeraccount VARCHAR(20), 
+    sequence INTEGER
+);
+
+INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
+
+-- Running upgrade 210693f3123d -> 54cde9847798
+
+ALTER TABLE cdr MODIFY accountcode VARCHAR(80) NULL;
+
+ALTER TABLE cdr MODIFY peeraccount VARCHAR(80) NULL;
+
+UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';
+

contrib/realtime/mysql/mysql_voicemail.sql

@@ -0,0 +1,35 @@
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> a2e9769475e
+
+CREATE TABLE voicemail_messages (
+    dir VARCHAR(255) NOT NULL, 
+    msgnum INTEGER NOT NULL, 
+    context VARCHAR(80), 
+    macrocontext VARCHAR(80), 
+    callerid VARCHAR(80), 
+    origtime INTEGER, 
+    duration INTEGER, 
+    recording BLOB, 
+    flag VARCHAR(30), 
+    category VARCHAR(30), 
+    mailboxuser VARCHAR(30), 
+    mailboxcontext VARCHAR(30), 
+    msg_id VARCHAR(40)
+);
+
+ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
+
+CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
+
+INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
+
+-- Running upgrade a2e9769475e -> 39428242f7f5
+
+ALTER TABLE voicemail_messages MODIFY recording BLOB(4294967295) NULL;
+
+UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
+

contrib/realtime/postgresql/postgresql_cdr.sql

@@ -0,0 +1,45 @@
+BEGIN;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> 210693f3123d
+
+CREATE TABLE cdr (
+    accountcode VARCHAR(20), 
+    src VARCHAR(80), 
+    dst VARCHAR(80), 
+    dcontext VARCHAR(80), 
+    clid VARCHAR(80), 
+    channel VARCHAR(80), 
+    dstchannel VARCHAR(80), 
+    lastapp VARCHAR(80), 
+    lastdata VARCHAR(80), 
+    start TIMESTAMP WITHOUT TIME ZONE, 
+    answer TIMESTAMP WITHOUT TIME ZONE, 
+    "end" TIMESTAMP WITHOUT TIME ZONE, 
+    duration INTEGER, 
+    billsec INTEGER, 
+    disposition VARCHAR(45), 
+    amaflags VARCHAR(45), 
+    userfield VARCHAR(256), 
+    uniqueid VARCHAR(150), 
+    linkedid VARCHAR(150), 
+    peeraccount VARCHAR(20), 
+    sequence INTEGER
+);
+
+INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
+
+-- Running upgrade 210693f3123d -> 54cde9847798
+
+ALTER TABLE cdr ALTER COLUMN accountcode TYPE VARCHAR(80);
+
+ALTER TABLE cdr ALTER COLUMN peeraccount TYPE VARCHAR(80);
+
+UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';
+
+COMMIT;
+

contrib/realtime/postgresql/postgresql_voicemail.sql

@@ -0,0 +1,39 @@
+BEGIN;
+
+CREATE TABLE alembic_version (
+    version_num VARCHAR(32) NOT NULL, 
+    CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
+);
+
+-- Running upgrade  -> a2e9769475e
+
+CREATE TABLE voicemail_messages (
+    dir VARCHAR(255) NOT NULL, 
+    msgnum INTEGER NOT NULL, 
+    context VARCHAR(80), 
+    macrocontext VARCHAR(80), 
+    callerid VARCHAR(80), 
+    origtime INTEGER, 
+    duration INTEGER, 
+    recording BYTEA, 
+    flag VARCHAR(30), 
+    category VARCHAR(30), 
+    mailboxuser VARCHAR(30), 
+    mailboxcontext VARCHAR(30), 
+    msg_id VARCHAR(40)
+);
+
+ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
+
+CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
+
+INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
+
+-- Running upgrade a2e9769475e -> 39428242f7f5
+
+ALTER TABLE voicemail_messages ALTER COLUMN recording TYPE BYTEA;
+
+UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
+
+COMMIT;
+

doc/CHANGES-staging/app_confbridge_hear_join.txt

@@ -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.

doc/CHANGES-staging/app_queue_music.txt

@@ -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.

doc/CHANGES-staging/chan_dahdi_cadences.txt

@@ -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.

doc/CHANGES-staging/chan_pjsip_180_sdp.txt

@@ -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.

doc/CHANGES-staging/chan_pjsip_flash.txt

@@ -1,4 +0,0 @@
-Subject: chan_pjsip
-
-Hook flash events can now be sent on a PJSIP channel
-if requested to do so.

doc/CHANGES-staging/chan_sip_session-timer_on_update.txt

@@ -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
-

doc/CHANGES-staging/cli_eval_function.txt

@@ -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.

doc/CHANGES-staging/func_db.txt

@@ -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.

doc/CHANGES-staging/func_evalexten.txt

@@ -1,4 +0,0 @@
-Subject: func_evalexten
-
-This adds the EVAL_EXTEN function which may be
-used to evaluate data at dialplan extensions.

doc/UPGRADE-staging/res_pjsip_async_operations.txt

@@ -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.

res/res_pjsip_transport_websocket.c

@@ -225,6 +225,7 @@ static int transport_create(void *data)
 	pj_sockaddr_parse(pj_AF_UNSPEC(), 0, pj_cstr(&buf, ws_addr_str), &newtransport->transport.local_addr);
 	pj_strdup2(pool, &newtransport->transport.local_name.host, ast_sockaddr_stringify_addr(ast_websocket_local_address(newtransport->ws_session)));
 	newtransport->transport.local_name.port = ast_sockaddr_port(ast_websocket_local_address(newtransport->ws_session));
+	pj_strdup2(pool, &newtransport->transport.remote_name.host, ast_sockaddr_stringify_addr(ast_websocket_remote_address(newtransport->ws_session)));
 
 	newtransport->transport.flag = pjsip_transport_get_flag_from_type((pjsip_transport_type_e)newtransport->transport.key.type);
 	newtransport->transport.dir = PJSIP_TP_DIR_INCOMING;