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asterisk-13.x: fix AST-2019-002 & AST-2019-003

Browse source 
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>
This commit is contained in:
Sebastian Kemper 2019-07-12 20:43:14 +02:00
parent a5b136487e
commit f4eaf906cc
3 changed files with 75 additions and 1 deletions
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2
net/asterisk-13.x/Makefile
View file

@ -11,7 +11,7 @@ include $(TOPDIR)/rules.mk
PKG_NAME:=asterisk13
PKG_VERSION:=13.24.0
PKG_RELEASE:=1
PKG_RELEASE:=2
PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases

35
net/asterisk-13.x/patches/110-AST-2019-002-13.diff Normal file
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@ -0,0 +1,35 @@
From a9d8b56831146166abc7fb8abe8ae8aaff295358 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
---
diff --git a/res/res_pjsip_messaging.c b/res/res_pjsip_messaging.c
index 10c5f29..76d37f2 100644
--- a/res/res_pjsip_messaging.c
+++ b/res/res_pjsip_messaging.c
@@ -91,10 +91,13 @@
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

39
net/asterisk-13.x/patches/120-AST-2019-003-13.diff Normal file
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@ -0,0 +1,39 @@
From 3ab9291a563656dfebcb7de67c86351541f3de1c 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 7c8928d..223ff3c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10911,7 +10911,13 @@
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;