Update for 16.4.1

Merge "res_pjsip_messaging: Check for body in in-dialog message" into 16.4
res_pjsip_messaging: Check for body in in-dialog message
2025-08-27 16:05:54 +00:00 · 2019-07-11 14:25:15 -05:00 · 2019-07-11 14:13:29 -05:00 · 2019-07-11 12:03:02 -06:00 · 2019-07-11 11:39:05 -06:00 · 2019-05-30 12:26:01 -05:00
20 changed files with 87360 additions and 41 deletions
--- a/.lastclean
+++ b/.lastclean
@@ -0,0 +1 @@
+40
--- a/.version
+++ b/.version
@@ -0,0 +1 @@
+16.4.1
--- a/45
+++ b/45
@@ -12,6 +12,51 @@
 ===
 ==============================================================================

+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 16.3.0 to Asterisk 16.4.0 ------------
+------------------------------------------------------------------------------
+
+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.
+
+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.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 16.2.0 to Asterisk 16.3.0 ----------
 ------------------------------------------------------------------------------
--- a/84458
+++ b/84458
--- a/asterisk-16.4.1-summary.html
+++ b/asterisk-16.4.1-summary.html
@@ -0,0 +1,15 @@
+<!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-16.4.1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-16.4.1</h3><h3 align="center">Date: 2019-07-11</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 has been made to address one or more security vulnerabilities that have been identified. A security advisory document has been published for each vulnerability that includes additional information. Users of versions of Asterisk that are affected are strongly encouraged to review the advisories and determine what action they should take to protect their systems from these issues.</p><p>Security Advisories:</p><ul>
+<li><a href="http://downloads.asterisk.org/pub/security/AST-2019-002,AST-2019-003.html">AST-2019-002,AST-2019-003</a></li>
+</ul><p>The data in this summary reflects changes that have been made since the previous release, asterisk-16.4.0.</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 Francesco Castellano <francesco.castellano@messagenet.it><br/>1 George Joseph <gjoseph@digium.com><br/></td><td width="33%"><td width="33%">1 Gil Richard<br/>1 Gil Richard <grichard@intertalksystems.com><br/>1 Francesco Castellano <francesco.castellano@messagenet.it><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>Security</h3><h4>Category: Channels/chan_sip/Interoperability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28465">ASTERISK-28465</a>: Broken SDP can cause a segfault in a T.38 reINVITE<br/>Reported by: Francesco Castellano<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3c185d0620910d8fded3f2baec64207d909e0fea">[3c185d0620]</a> Francesco Castellano -- chan_sip: Handle invalid SDP answer to T.38 re-invite</li>
+</ul><br><h4>Category: Resources/res_pjsip_messaging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28447">ASTERISK-28447</a>: res_pjsip_messaging: In-dialog MESSAGE with no body causes crash<br/>Reported by: Gil Richard<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c2319178b2d789a45994d25c463118d9f4ee5cea">[c2319178b2]</a> George Joseph -- res_pjsip_messaging:  Check for body in in-dialog message</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>
--- a/asterisk-16.4.1-summary.txt
+++ b/asterisk-16.4.1-summary.txt
@@ -0,0 +1,93 @@
+                                Release Summary
+
+                                asterisk-16.4.1
+
+                                Date: 2019-07-11
+
+                           <asteriskteam@digium.com>
+
+     ----------------------------------------------------------------------
+
+                               Table of Contents
+
+    1. Summary
+    2. Contributors
+    3. Closed Issues
+    4. Diffstat
+
+     ----------------------------------------------------------------------
+
+                                    Summary
+
+                                 [Back to Top]
+
+   This release has been made to address one or more security vulnerabilities
+   that have been identified. A security advisory document has been published
+   for each vulnerability that includes additional information. Users of
+   versions of Asterisk that are affected are strongly encouraged to review
+   the advisories and determine what action they should take to protect their
+   systems from these issues.
+
+   Security Advisories:
+
+     * AST-2019-002,AST-2019-003
+
+   The data in this summary reflects changes that have been made since the
+   previous release, asterisk-16.4.0.
+
+     ----------------------------------------------------------------------
+
+                                  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 Francesco Castellano                            1 Gil Richard            
+   1 George Joseph                                   1 Gil Richard            
+                                                     1 Francesco Castellano   
+
+     ----------------------------------------------------------------------
+
+                                 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.
+
+  Security
+
+    Category: Channels/chan_sip/Interoperability
+
+   ASTERISK-28465: Broken SDP can cause a segfault in a T.38 reINVITE
+   Reported by: Francesco Castellano
+     * [3c185d0620] Francesco Castellano -- chan_sip: Handle invalid SDP
+       answer to T.38 re-invite
+
+    Category: Resources/res_pjsip_messaging
+
+   ASTERISK-28447: res_pjsip_messaging: In-dialog MESSAGE with no body causes
+   crash
+   Reported by: Gil Richard
+     * [c2319178b2] George Joseph -- res_pjsip_messaging: Check for body in
+       in-dialog message
+
+     ----------------------------------------------------------------------
+
+                                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
--- a/channels/chan_dahdi.h
+++ b/channels/chan_dahdi.h
@@ -39,6 +39,7 @@
 #include "asterisk/channel.h"
 #include "asterisk/dsp.h"
 #include "asterisk/app.h"
+#include "asterisk/mwi.h"

 #if defined(__cplusplus) || defined(c_plusplus)
 extern "C" {
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10965,7 +10965,13 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 			    ast_rtp_lookup_mime_multiple2(s3, NULL, newnoncodeccapability, 0, 0));
 	}

-	if (portno != -1 || vportno != -1 || tportno != -1) {
+	/* When UDPTL is negotiated it is expected that there are no compatible codecs as audio or
+	 * video is not being transported, thus we continue in this function further up if that is
+	 * the case. If we receive an SDP answer containing both a UDPTL stream and another media
+	 * stream however we need to check again to ensure that there is at least one joint codec
+	 * instead of assuming there is one.
+	 */
+	if ((portno != -1 || vportno != -1 || tportno != -1) && ast_format_cap_count(newjointcapability)) {
 		/* We are now ready to change the sip session and RTP structures with the offered codecs, since
 		   they are acceptable */
 		unsigned int framing;
--- a/channels/sig_pri.c
+++ b/channels/sig_pri.c
@@ -101,6 +101,7 @@
 #include "asterisk/options.h"
 #include "asterisk/pbx.h"
 #include "asterisk/app.h"
+#include "asterisk/mwi.h"
 #include "asterisk/file.h"
 #include "asterisk/callerid.h"
 #include "asterisk/say.h"
--- a/contrib/realtime/mysql/mysql_cdr.sql
+++ b/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';
+
--- a/contrib/realtime/mysql/mysql_config.sql
+++ b/contrib/realtime/mysql/mysql_config.sql
--- a/contrib/realtime/mysql/mysql_voicemail.sql
+++ b/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';
+
--- a/contrib/realtime/postgresql/postgresql_cdr.sql
+++ b/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;
+
--- a/contrib/realtime/postgresql/postgresql_config.sql
+++ b/contrib/realtime/postgresql/postgresql_config.sql
--- a/contrib/realtime/postgresql/postgresql_voicemail.sql
+++ b/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;
+
--- a/doc/CHANGES-staging/app_confbridge_remb_behavior_all.txt
+++ b/doc/CHANGES-staging/app_confbridge_remb_behavior_all.txt
@@ -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..
--- a/doc/CHANGES-staging/app_dial_ringtime_progresstime.txt
+++ b/doc/CHANGES-staging/app_dial_ringtime_progresstime.txt
@@ -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
--- a/doc/CHANGES-staging/res_pjsip_add_norefersub_global_config.txt
+++ b/doc/CHANGES-staging/res_pjsip_add_norefersub_global_config.txt
@@ -1,13 +0,0 @@
-res_pjsip:  Added a norefersub configuration setting
-
-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.
--- a/doc/CHANGES-staging/rtp_ice_include_local_address.txt
+++ b/doc/CHANGES-staging/rtp_ice_include_local_address.txt
@@ -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.
--- a/res/res_pjsip_messaging.c
+++ b/res/res_pjsip_messaging.c
@@ -90,10 +90,13 @@ static enum pjsip_status_code check_content_type_in_dialog(const pjsip_rx_data *
 	static const pj_str_t text = { "text", 4};
 	static const pj_str_t application = { "application", 11};

+	if (!(rdata->msg_info.msg->body && rdata->msg_info.msg->body->len > 0)) {
+		return res;
+	}
+
 	/* We'll accept any text/ or application/ content type */
-	if (rdata->msg_info.msg->body && rdata->msg_info.msg->body->len
-		&& (pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &text) == 0
-			|| pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &application) == 0)) {
+	if (pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &text) == 0
+			|| pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &application) == 0) {
 		res = PJSIP_SC_OK;
 	} else if (rdata->msg_info.ctype
 		&& (pj_stricmp(&rdata->msg_info.ctype->media.type, &text) == 0
Author	SHA1	Message	Date
Asterisk Development Team	79e42ad28b	Update for 16.4.1	2019-07-11 14:25:15 -05:00
Benjamin Keith Ford	aefe8cdf11	Merge "res_pjsip_messaging: Check for body in in-dialog message" into 16.4	2019-07-11 14:13:29 -05:00
George Joseph	c2319178b2	res_pjsip_messaging: Check for body in in-dialog message We now check that a body exists and it has a length > 0 before attempting to process it. ASTERISK-28447 Reported-by: Gil Richard Change-Id: Ic469544b22ab848734636588d4c93426cc6f4b1f	2019-07-11 12:03:02 -06:00
Francesco Castellano	3c185d0620	chan_sip: Handle invalid SDP answer to T.38 re-invite The chan_sip module performs a T.38 re-invite using a single media stream of udptl, and expects the SDP answer to be the same. If an SDP answer is received instead that contains an additional media stream with no joint codec a crash will occur as the code assumes that at least one joint codec will exist in this scenario. This change removes this assumption. ASTERISK-28465 Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87	2019-07-11 11:39:05 -06:00
Asterisk Development Team	a069c71f7f	Update for 16.4.0	2019-05-30 12:26:01 -05:00
Asterisk Development Team	05f2b24110	Update CHANGES and UPGRADE.txt for 16.4.0	2019-05-30 12:09:49 -05:00
Friendly Automation	745a9681f0	Merge "build: Fix file format in CHANGES-staging." into 16.4	2019-05-30 05:56:49 -05:00
Ben Ford	f23d0002c6	build: Fix file format in CHANGES-staging. One of the change files doesn't conform to the format that the release scripts need in order to parse it. Change-Id: Ie0b634cf27e4cbc671b9fe92993b6f2ecf60254c	2019-05-24 08:30:40 -06:00
Guido Falsi	e1977f4e85	chan_dahdi: add missing include. After some definitions have been moved to asterisk/mwi.h the files channels/chan_dahdi.h channels/sig_pri.c are missing this new include. ASTERISK-28427 #close Change-Id: Ia8cc595eeda653324643f40dcd9799d4c3f0ac91	2019-05-23 08:50:16 -06:00
Asterisk Development Team	5af18dc5e3	Update for 16.4.0-rc1	2019-05-22 13:41:45 -05:00