asterisk-16.x: fix AST-2019-002 & AST-2019-003

https://downloads.asterisk.org/pub/security/AST-2019-002.html
https://downloads.asterisk.org/pub/security/AST-2019-003.html

Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>

diff --git a/net/asterisk-16.x/Makefile b/net/asterisk-16.x/Makefile
index 00da40fca3eb754b4a283dacf0215b78b36db6dd..15588625e29ebd4fdbdffbcfff5ddd228f661f43 100644 (file)
--- a/net/asterisk-16.x/Makefile
+++ b/net/asterisk-16.x/Makefile
@@ -10,7 +10,7 @@ include $(TOPDIR)/rules.mk
 AST_MAJOR_VERSION:=16
 PKG_NAME:=asterisk$(AST_MAJOR_VERSION)
 PKG_VERSION:=$(AST_MAJOR_VERSION).3.0
-PKG_RELEASE:=3
+PKG_RELEASE:=4
 
 PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
 PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases

diff --git a/net/asterisk-16.x/patches/140-AST-2019-002-16.diff b/net/asterisk-16.x/patches/140-AST-2019-002-16.diff
new file mode 100644 (file)
index 0000000..635d837
--- /dev/null
+++ b/net/asterisk-16.x/patches/140-AST-2019-002-16.diff
@@ -0,0 +1,40 @@
+From 785bf3a755e47d92caef110e6040295764d08127 Mon Sep 17 00:00:00 2001
+From: George Joseph <gjoseph@digium.com>
+Date: Wed, 12 Jun 2019 12:03:04 -0600
+Subject: [PATCH] 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
+---
+ res/res_pjsip_messaging.c | 9 ++++++---
+ 1 file changed, 6 insertions(+), 3 deletions(-)
+
+diff --git a/res/res_pjsip_messaging.c b/res/res_pjsip_messaging.c
+index 0e10a8f047..930cf84a53 100644
+--- 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
+-- 
+2.21.0
+

diff --git a/net/asterisk-16.x/patches/150-AST-2019-003-16.diff b/net/asterisk-16.x/patches/150-AST-2019-003-16.diff
new file mode 100644 (file)
index 0000000..90b9d5d
--- /dev/null
+++ b/net/asterisk-16.x/patches/150-AST-2019-003-16.diff
@@ -0,0 +1,39 @@
+From 1e4df0215af4f192ed06a7fc7589c799f1ec6091 Mon Sep 17 00:00:00 2001
+From: Francesco Castellano <francesco.castellano@messagenet.it>
+Date: Fri, 28 Jun 2019 18:15:31 +0200
+Subject: [PATCH] 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
+---
+
+diff --git a/channels/chan_sip.c b/channels/chan_sip.c
+index 898b646..a609ff8 100644
+--- a/channels/chan_sip.c
++++ b/channels/chan_sip.c
+@@ -10965,7 +10965,13 @@
+                           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;