Merge pull request #377 from dhewg/pull/asterisk

asterisk related updates
This commit is contained in:
Jiri Slachta 2018-10-15 11:41:27 +02:00 committed by GitHub
commit 72cb5b8350
No known key found for this signature in database
GPG key ID: 4AEE18F83AFDEB23
20 changed files with 37 additions and 484 deletions

View file

@ -11,7 +11,7 @@ include $(TOPDIR)/rules.mk
PKG_NAME:=pjproject
PKG_VERSION:=2.7.2
PKG_RELEASE:=2
PKG_RELEASE:=3
PKG_SOURCE:=pjproject-$(PKG_VERSION).tar.bz2
PKG_SOURCE_URL:=http://www.pjsip.org/release/$(PKG_VERSION)

View file

@ -1,6 +1,6 @@
--- /dev/null
+++ b/pjlib/include/pj/config_site.h
@@ -0,0 +1,92 @@
@@ -0,0 +1,79 @@
+/*
+ * Asterisk config_site.h
+ */
@ -8,37 +8,16 @@
+#include <sys/select.h>
+
+/*
+ * Since both pjproject and asterisk source files will include config_site.h,
+ * we need to make sure that only pjproject source files include asterisk_malloc_debug.h.
+ */
+
+/* #if defined(MALLOC_DEBUG) && !defined(_ASTERISK_ASTMM_H)
+ * #include "asterisk_malloc_debug.h"
+ * #endif
+ */
+
+/*
+ * Defining PJMEDIA_HAS_SRTP to 0 does NOT disable Asterisk's ability to use srtp.
+ * It only disables the pjmedia srtp transport which Asterisk doesn't use.
+ * The reason for the disable is that while Asterisk works fine with older libsrtp
+ * versions, newer versions of pjproject won't compile with them.
+ */
+
+/*
+ * This doesn't disable SRTP completely, so we have to keep using the external
+ * libsrtp, otherwise pjsip would just build the internal one.
+ */
+
+#define PJMEDIA_HAS_SRTP 0
+
+/*
+ * Defining PJMEDIA_HAS_WEBRTC_AEC to 0 does NOT disable Asterisk's ability to use
+ * webrtc. It only disables the pjmedia webrtc transport which Asterisk doesn't use.
+ */
+#define PJMEDIA_HAS_WEBRTC_AEC 0
+
+#define PJ_HAS_IPV6 1
+#define NDEBUG 1
+
+#define PJ_MAX_HOSTNAME (256)
+#define PJSIP_MAX_URL_SIZE (512)
+#ifdef PJ_HAS_LINUX_EPOLL
@ -79,7 +58,7 @@
+ Enabling it will result in SEGFAULTS when URIs containing escape sequences are encountered.
+*/
+#undef PJSIP_UNESCAPE_IN_PLACE
+#define PJSIP_MAX_PKT_LEN 6000
+#define PJSIP_MAX_PKT_LEN 32000
+
+#undef PJ_TODO
+#define PJ_TODO(x)
@ -93,3 +72,11 @@
+#define PJMEDIA_MAX_SDP_BANDW 4
+#define PJMEDIA_MAX_SDP_ATTR (PJMEDIA_MAX_SDP_FMT*2 + 4)
+#define PJMEDIA_MAX_SDP_MEDIA 16
+
+/*
+ * Turn off the periodic sending of CRLNCRLN. Default is on (90 seconds),
+ * which conflicts with the global section's keep_alive_interval option in
+ * pjsip.conf.
+ */
+#define PJSIP_TCP_KEEP_ALIVE_INTERVAL 0
+#define PJSIP_TLS_KEEP_ALIVE_INTERVAL 0

View file

@ -10,12 +10,12 @@
include $(TOPDIR)/rules.mk
PKG_NAME:=asterisk13
PKG_VERSION:=13.20.0
PKG_RELEASE:=4
PKG_VERSION:=13.23.1
PKG_RELEASE:=1
PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases
PKG_HASH:=a3d6d953f844867ea11e0be22ee6225049cd4f5870df6ab23454623bcfbc94d5
PKG_HASH:=c772acbfdddb9250bfe07f7e20a7efb6a79a6c123832727429486c78d44fc78c
PKG_BUILD_DIR:=$(BUILD_DIR)/asterisk-$(PKG_VERSION)
PKG_BUILD_DEPENDS:=libxml2/host
@ -425,7 +425,7 @@ define Package/asterisk13
$(call Package/asterisk13/Default)
TITLE:=Complete open source PBX, v$(PKG_VERSION)
MENU:=1
DEPENDS:=$(CXX_DEPENDS) +jansson +libcap +libedit +libncurses +libopenssl +libsqlite3 +libuuid +libxml2 +zlib
DEPENDS:=$(CXX_DEPENDS) +jansson +libcap +libedit +libopenssl +libsqlite3 +libuuid +libxml2 +zlib
endef
define Package/asterisk13/description
@ -532,7 +532,6 @@ CONFIGURE_ARGS+= \
--without-nbs \
--with-libedit="$(STAGING_DIR)/usr" \
--with-libxml2 \
--with-ncurses="$(STAGING_DIR)/usr" \
$(if $(CONFIG_PACKAGE_$(PKG_NAME)-res-snmp),--with-netsnmp="$(STAGING_DIR)/usr",--without-netsnmp) \
--without-newt \
$(if $(CONFIG_PACKAGE_$(PKG_NAME)-format-ogg-vorbis),--with-ogg="$(STAGING_DIR)/usr",--without-ogg) \
@ -541,7 +540,6 @@ CONFIGURE_ARGS+= \
$(if $(CONFIG_PACKAGE_$(PKG_NAME)-pgsql),--with-postgres="$(STAGING_DIR)/usr",--without-postgres) \
$(if $(CONFIG_PACKAGE_$(PKG_NAME)-util-smsq),--with-popt="$(STAGING_DIR)/usr",--without-popt) \
$(if $(CONFIG_PACKAGE_$(PKG_NAME)-chan-console),--with-portaudio="$(STAGING_DIR)/usr",--without-portaudio) \
--without-pwlib \
--without-radius \
$(if $(CONFIG_PACKAGE_$(PKG_NAME)-res-fax-spandsp),--with-spandsp="$(STAGING_DIR)/usr",--without-spandsp) \
--without-sdl \

View file

@ -1,6 +1,6 @@
--- a/configure.ac
+++ b/configure.ac
@@ -953,19 +953,6 @@ AC_LINK_IFELSE(
@@ -1016,19 +1016,6 @@ AC_LINK_IFELSE(
]
)

View file

@ -1,6 +1,6 @@
--- a/configure.ac
+++ b/configure.ac
@@ -1287,7 +1287,6 @@ AC_LINK_IFELSE(
@@ -1363,7 +1363,6 @@ AC_LINK_IFELSE(
#include <resolv.h>],
[int foo = res_ninit(NULL);])],
AC_MSG_RESULT(yes)

View file

@ -1,6 +1,6 @@
--- a/res/ael/ael.flex
+++ b/res/ael/ael.flex
@@ -79,6 +79,12 @@
@@ -79,6 +79,12 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revisi
#if !defined(GLOB_ABORTED)
#define GLOB_ABORTED GLOB_ABEND
#endif
@ -13,10 +13,9 @@
#include "asterisk/logger.h"
#include "asterisk/utils.h"
Only in asterisk-11.7.0: res/ael/ael.tab.o
--- a/res/ael/ael_lex.c
+++ b/res/ael/ael_lex.c
@@ -838,6 +838,12 @@
@@ -838,6 +838,12 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revisi
#if !defined(GLOB_ABORTED)
#define GLOB_ABORTED GLOB_ABEND
#endif

View file

@ -1,6 +1,6 @@
--- a/configure.ac
+++ b/configure.ac
@@ -170,6 +170,9 @@ case "${host_os}" in
@@ -175,6 +175,9 @@ case "${host_os}" in
linux-gnu*)
OSARCH=linux-gnu
;;
@ -10,9 +10,9 @@
kfreebsd*-gnu)
OSARCH=kfreebsd-gnu
;;
@@ -1411,9 +1414,11 @@ if test "${PBX_BFD}" = "0"; then
AST_EXT_LIB_CHECK([BFD], [bfd], [bfd_check_format], [bfd.h], [-ldl -liberty -lz])
fi
@@ -1495,9 +1498,11 @@ AST_EXT_LIB_CHECK([BFD], [bfd], [bfd_ope
# openSUSE requires -lz
AST_EXT_LIB_CHECK([BFD], [bfd], [bfd_openr], [bfd.h], [-ldl -liberty -lz])
-if test "x${OSARCH}" = "xlinux-gnu" ; then
+case "${OSARCH}" in

View file

@ -1,7 +1,7 @@
--- a/utils/Makefile
+++ b/utils/Makefile
@@ -179,7 +179,6 @@ check_expr2: $(ASTTOPDIR)/main/ast_expr2
$(CC) -g -o check_expr2 ast_expr2fz.o ast_expr2z.o -lm
@@ -180,7 +180,6 @@ check_expr2: $(ASTTOPDIR)/main/ast_expr2
$(CC) -g -o check_expr2 ast_expr2fz.o ast_expr2z.o -lm $(_ASTLDFLAGS)
$(ECHO_PREFIX) echo " [RM] ast_expr2fz.o ast_expr2z.o"
rm ast_expr2z.o ast_expr2fz.o
- ./check_expr2 expr2.testinput

View file

@ -17,7 +17,7 @@
* build.h
--- a/Makefile
+++ b/Makefile
@@ -480,7 +480,7 @@ doc/core-en_US.xml: makeopts .lastclean
@@ -488,7 +488,7 @@ doc/core-en_US.xml: makeopts .lastclean
@echo "<docs xmlns:xi=\"http://www.w3.org/2001/XInclude\">" >> $@
@for x in $(MOD_SUBDIRS); do \
printf "$$x " ; \

View file

@ -1,101 +0,0 @@
From 4eeb16d1a316aa3d6f5710a2f6beffb0fecb6121 Mon Sep 17 00:00:00 2001
From: Richard Mudgett <rmudgett@digium.com>
Date: Mon, 30 Apr 2018 17:38:58 -0500
Subject: [PATCH] AST-2018-008: Fix enumeration of endpoints from ACL rejected addresses.
When endpoint specific ACL rules block a SIP request they respond with a
403 forbidden. However, if an endpoint is not identified then a 401
unauthorized response is sent. This vulnerability just discloses which
requests hit a defined endpoint. The ACL rules cannot be bypassed to gain
access to the disclosed endpoints.
* Made endpoint specific ACL rules now respond with a 401 unauthorized
which is the same as if an endpoint were not identified. The fix is
accomplished by replacing the found endpoint with the artificial endpoint
which always fails authentication.
ASTERISK-27818
Change-Id: Icb275a54ff8e2df6c671a6d9bda37b5d732b3b32
---
diff --git a/res/res_pjsip/pjsip_distributor.c b/res/res_pjsip/pjsip_distributor.c
index e056b60..19266df 100644
--- a/res/res_pjsip/pjsip_distributor.c
+++ b/res/res_pjsip/pjsip_distributor.c
@@ -666,6 +666,26 @@
ao2_unlock(unid);
}
+static int apply_endpoint_acl(pjsip_rx_data *rdata, struct ast_sip_endpoint *endpoint);
+static int apply_endpoint_contact_acl(pjsip_rx_data *rdata, struct ast_sip_endpoint *endpoint);
+
+static void apply_acls(pjsip_rx_data *rdata)
+{
+ struct ast_sip_endpoint *endpoint;
+
+ /* Is the endpoint allowed with the source or contact address? */
+ endpoint = rdata->endpt_info.mod_data[endpoint_mod.id];
+ if (endpoint != artificial_endpoint
+ && (apply_endpoint_acl(rdata, endpoint)
+ || apply_endpoint_contact_acl(rdata, endpoint))) {
+ ast_debug(1, "Endpoint '%s' not allowed by ACL\n",
+ ast_sorcery_object_get_id(endpoint));
+
+ /* Replace the rdata endpoint with the artificial endpoint. */
+ ao2_replace(rdata->endpt_info.mod_data[endpoint_mod.id], artificial_endpoint);
+ }
+}
+
static pj_bool_t endpoint_lookup(pjsip_rx_data *rdata)
{
struct ast_sip_endpoint *endpoint;
@@ -684,6 +704,7 @@
ao2_unlink(unidentified_requests, unid);
ao2_ref(unid, -1);
}
+ apply_acls(rdata);
return PJ_FALSE;
}
@@ -743,6 +764,8 @@
ast_sip_report_invalid_endpoint(name, rdata);
}
}
+
+ apply_acls(rdata);
return PJ_FALSE;
}
@@ -826,16 +849,11 @@
ast_assert(endpoint != NULL);
- if (endpoint!=artificial_endpoint) {
- if (apply_endpoint_acl(rdata, endpoint) || apply_endpoint_contact_acl(rdata, endpoint)) {
- if (!is_ack) {
- pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 403, NULL, NULL, NULL);
- }
- return PJ_TRUE;
- }
+ if (is_ack) {
+ return PJ_FALSE;
}
- if (!is_ack && ast_sip_requires_authentication(endpoint, rdata)) {
+ if (ast_sip_requires_authentication(endpoint, rdata)) {
pjsip_tx_data *tdata;
struct unidentified_request *unid;
@@ -871,6 +889,10 @@
return PJ_TRUE;
}
pjsip_tx_data_dec_ref(tdata);
+ } else if (endpoint == artificial_endpoint) {
+ /* Uh. Oh. The artificial endpoint couldn't challenge so block the request. */
+ pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 500, NULL, NULL, NULL);
+ return PJ_TRUE;
}
return PJ_FALSE;

View file

@ -1,89 +0,0 @@
From e6b0c4d27e0392a7b4b4b6717a6d1e0ea049b550 Mon Sep 17 00:00:00 2001
From: Sean Bright <sean.bright@gmail.com>
Date: Thu, 16 Aug 2018 11:45:53 -0400
Subject: [PATCH] AST-2018-009: Fix crash processing websocket HTTP Upgrade
requests
The HTTP request processing in res_http_websocket allocates additional
space on the stack for various headers received during an Upgrade request.
An attacker could send a specially crafted request that causes this code
to overflow the stack, resulting in a crash.
* No longer allocate memory from the stack in a loop to parse the header
values. NOTE: There is a slight API change when using the passed in
strings as is. We now require the passed in strings to no longer have
leading or trailing whitespace. This isn't a problem as the only callers
have already done this before passing the strings to the affected
function.
ASTERISK-28013 #close
Change-Id: Ia564825a8a95e085fd17e658cb777fe1afa8091a
---
res/res_http_websocket.c | 25 ++++++++++++++-----------
1 file changed, 14 insertions(+), 11 deletions(-)
diff --git a/res/res_http_websocket.c b/res/res_http_websocket.c
index 440bf41..0ff876b 100644
--- a/res/res_http_websocket.c
+++ b/res/res_http_websocket.c
@@ -736,7 +736,8 @@ static void websocket_bad_request(struct ast_tcptls_session_instance *ser)
int AST_OPTIONAL_API_NAME(ast_websocket_uri_cb)(struct ast_tcptls_session_instance *ser, const struct ast_http_uri *urih, const char *uri, enum ast_http_method method, struct ast_variable *get_vars, struct ast_variable *headers)
{
struct ast_variable *v;
- char *upgrade = NULL, *key = NULL, *key1 = NULL, *key2 = NULL, *protos = NULL, *requested_protocols = NULL, *protocol = NULL;
+ const char *upgrade = NULL, *key = NULL, *key1 = NULL, *key2 = NULL, *protos = NULL;
+ char *requested_protocols = NULL, *protocol = NULL;
int version = 0, flags = 1;
struct ast_websocket_protocol *protocol_handler = NULL;
struct ast_websocket *session;
@@ -755,16 +756,15 @@ int AST_OPTIONAL_API_NAME(ast_websocket_uri_cb)(struct ast_tcptls_session_instan
/* Get the minimum headers required to satisfy our needs */
for (v = headers; v; v = v->next) {
if (!strcasecmp(v->name, "Upgrade")) {
- upgrade = ast_strip(ast_strdupa(v->value));
+ upgrade = v->value;
} else if (!strcasecmp(v->name, "Sec-WebSocket-Key")) {
- key = ast_strip(ast_strdupa(v->value));
+ key = v->value;
} else if (!strcasecmp(v->name, "Sec-WebSocket-Key1")) {
- key1 = ast_strip(ast_strdupa(v->value));
+ key1 = v->value;
} else if (!strcasecmp(v->name, "Sec-WebSocket-Key2")) {
- key2 = ast_strip(ast_strdupa(v->value));
+ key2 = v->value;
} else if (!strcasecmp(v->name, "Sec-WebSocket-Protocol")) {
- requested_protocols = ast_strip(ast_strdupa(v->value));
- protos = ast_strdupa(requested_protocols);
+ protos = v->value;
} else if (!strcasecmp(v->name, "Sec-WebSocket-Version")) {
if (sscanf(v->value, "%30d", &version) != 1) {
version = 0;
@@ -778,7 +778,7 @@ int AST_OPTIONAL_API_NAME(ast_websocket_uri_cb)(struct ast_tcptls_session_instan
ast_sockaddr_stringify(&ser->remote_address));
ast_http_error(ser, 426, "Upgrade Required", NULL);
return 0;
- } else if (ast_strlen_zero(requested_protocols)) {
+ } else if (ast_strlen_zero(protos)) {
/* If there's only a single protocol registered, and the
* client doesn't specify what protocol it's using, go ahead
* and accept the connection */
@@ -799,9 +799,12 @@ int AST_OPTIONAL_API_NAME(ast_websocket_uri_cb)(struct ast_tcptls_session_instan
return 0;
}
- /* Iterate through the requested protocols trying to find one that we have a handler for */
- while (!protocol_handler && (protocol = strsep(&requested_protocols, ","))) {
- protocol_handler = ao2_find(server->protocols, ast_strip(protocol), OBJ_KEY);
+ if (!protocol_handler && protos) {
+ requested_protocols = ast_strdupa(protos);
+ /* Iterate through the requested protocols trying to find one that we have a handler for */
+ while (!protocol_handler && (protocol = strsep(&requested_protocols, ","))) {
+ protocol_handler = ao2_find(server->protocols, ast_strip(protocol), OBJ_KEY);
+ }
}
/* If no protocol handler exists bump this back to the requester */
--
2.7.4

View file

@ -8,12 +8,12 @@
include $(TOPDIR)/rules.mk
PKG_NAME:=asterisk15
PKG_VERSION:=15.3.0
PKG_RELEASE:=4
PKG_VERSION:=15.6.1
PKG_RELEASE:=1
PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases
PKG_HASH:=f424f89f23b72f267ff9baab82d449bebbbf00c54e54fcd06b8fca13788b012c
PKG_HASH:=9cb86585fb4efcd86423fa7586e55d9d3c74fda43a2e0b1b9c9eb3742df73155
PKG_BUILD_DIR:=$(BUILD_DIR)/asterisk-$(PKG_VERSION)
PKG_BUILD_DEPENDS:=libxml2/host
@ -430,7 +430,7 @@ define Package/asterisk15
$(call Package/asterisk15/Default)
TITLE:=Complete open source PBX, v$(PKG_VERSION)
MENU:=1
DEPENDS:=$(CXX_DEPENDS) +jansson +libcap +libedit +libncurses +libopenssl +libsqlite3 +libuuid +libxml2 +zlib
DEPENDS:=$(CXX_DEPENDS) +jansson +libcap +libedit +libopenssl +libsqlite3 +libuuid +libxml2 +zlib
endef
define Package/asterisk15/description
@ -538,7 +538,6 @@ CONFIGURE_ARGS+= \
--without-pjproject-bundled \
--with-libedit="$(STAGING_DIR)/usr" \
--with-libxml2 \
--with-ncurses="$(STAGING_DIR)/usr" \
$(if $(CONFIG_PACKAGE_$(PKG_NAME)-res-snmp),--with-netsnmp="$(STAGING_DIR)/usr",--without-netsnmp) \
--without-newt \
--without-osptk \
@ -546,7 +545,6 @@ CONFIGURE_ARGS+= \
$(if $(CONFIG_PACKAGE_$(PKG_NAME)-pgsql),--with-postgres="$(STAGING_DIR)/usr",--without-postgres) \
$(if $(CONFIG_PACKAGE_$(PKG_NAME)-util-smsq),--with-popt="$(STAGING_DIR)/usr",--without-popt) \
$(if $(CONFIG_PACKAGE_$(PKG_NAME)-chan-console),--with-portaudio="$(STAGING_DIR)/usr",--without-portaudio) \
--without-pwlib \
--without-radius \
$(if $(CONFIG_PACKAGE_$(PKG_NAME)-res-fax-spandsp),--with-spandsp="$(STAGING_DIR)/usr",--without-spandsp) \
--without-sdl \

View file

@ -1,6 +1,6 @@
--- a/configure.ac
+++ b/configure.ac
@@ -985,19 +985,6 @@ AC_LINK_IFELSE(
@@ -1048,19 +1048,6 @@ AC_LINK_IFELSE(
]
)

View file

@ -1,6 +1,6 @@
--- a/configure.ac
+++ b/configure.ac
@@ -1319,7 +1319,6 @@ AC_LINK_IFELSE(
@@ -1395,7 +1395,6 @@ AC_LINK_IFELSE(
#include <resolv.h>],
[int foo = res_ninit(NULL);])],
AC_MSG_RESULT(yes)

View file

@ -1,6 +1,6 @@
--- a/configure.ac
+++ b/configure.ac
@@ -2709,7 +2709,7 @@ else
@@ -2790,7 +2790,7 @@ else
fi
# conf_bridge (binaural rendering): check if HRIRs are available

View file

@ -1,7 +1,7 @@
--- a/utils/Makefile
+++ b/utils/Makefile
@@ -182,7 +182,6 @@ check_expr2: $(ASTTOPDIR)/main/ast_expr2
$(CC) -g -o check_expr2 ast_expr2fz.o ast_expr2z.o -lm
@@ -183,7 +183,6 @@ check_expr2: $(ASTTOPDIR)/main/ast_expr2
$(CC) -g -o check_expr2 ast_expr2fz.o ast_expr2z.o -lm $(_ASTLDFLAGS)
$(ECHO_PREFIX) echo " [RM] ast_expr2fz.o ast_expr2z.o"
rm ast_expr2z.o ast_expr2fz.o
- ./check_expr2 expr2.testinput

View file

@ -17,7 +17,7 @@
* build.h
--- a/Makefile
+++ b/Makefile
@@ -479,7 +479,7 @@ doc/core-en_US.xml: makeopts .lastclean
@@ -487,7 +487,7 @@ doc/core-en_US.xml: makeopts .lastclean
@echo "<docs xmlns:xi=\"http://www.w3.org/2001/XInclude\">" >> $@
@for x in $(MOD_SUBDIRS); do \
printf "$$x " ; \

View file

@ -1,49 +0,0 @@
From 380b5ae0a1e4a68bfb098319a7ab86d3d34c2fcb Mon Sep 17 00:00:00 2001
From: Sean Bright <sean.bright@gmail.com>
Date: Mon, 16 Apr 2018 15:13:58 -0400
Subject: [PATCH] AST-2018-007: iostreams potential DoS when client connection closed prematurely
Before Asterisk sends an HTTP response (at least in the case of errors),
it attempts to read & discard the content of the request. If the client
lies about the Content-Length, or the connection is closed from the
client side before "Content-Length" bytes are sent, the request handling
thread will busy loop.
ASTERISK-27807
Change-Id: I945c5fc888ed92be625b8c35039fc6d2aa89c762
---
diff --git a/main/iostream.c b/main/iostream.c
index 4cddd43..20188cb 100644
--- a/main/iostream.c
+++ b/main/iostream.c
@@ -197,11 +197,18 @@
}
}
break;
+ case SSL_ERROR_SYSCALL:
+ /* Some non-recoverable I/O error occurred. The OpenSSL error queue may
+ * contain more information on the error. For socket I/O on Unix systems,
+ * consult errno for details. */
+ ast_debug(1, "TLS non-recoverable I/O error occurred: %s, %s\n", ERR_error_string(sslerr, err),
+ ssl_error_to_string(sslerr, res));
+ return -1;
default:
/* Report EOF for an undecoded SSL or transport error. */
ast_debug(1, "TLS transport or SSL error reading data: %s, %s\n", ERR_error_string(sslerr, err),
ssl_error_to_string(sslerr, res));
- return 0;
+ return -1;
}
if (!ms) {
/* Report EOF for a timeout */
@@ -317,7 +324,7 @@
while (remaining) {
ret = ast_iostream_read(stream, buf, remaining > sizeof(buf) ? sizeof(buf) : remaining);
- if (ret < 0) {
+ if (ret <= 0) {
return ret;
}
remaining -= ret;

View file

@ -1,101 +0,0 @@
From f597032e833a4d3e8e710e5b1416ba780f002b8b Mon Sep 17 00:00:00 2001
From: Richard Mudgett <rmudgett@digium.com>
Date: Mon, 30 Apr 2018 17:38:58 -0500
Subject: [PATCH] AST-2018-008: Fix enumeration of endpoints from ACL rejected addresses.
When endpoint specific ACL rules block a SIP request they respond with a
403 forbidden. However, if an endpoint is not identified then a 401
unauthorized response is sent. This vulnerability just discloses which
requests hit a defined endpoint. The ACL rules cannot be bypassed to gain
access to the disclosed endpoints.
* Made endpoint specific ACL rules now respond with a 401 unauthorized
which is the same as if an endpoint were not identified. The fix is
accomplished by replacing the found endpoint with the artificial endpoint
which always fails authentication.
ASTERISK-27818
Change-Id: Icb275a54ff8e2df6c671a6d9bda37b5d732b3b32
---
diff --git a/res/res_pjsip/pjsip_distributor.c b/res/res_pjsip/pjsip_distributor.c
index 51b95a2..0af447d 100644
--- a/res/res_pjsip/pjsip_distributor.c
+++ b/res/res_pjsip/pjsip_distributor.c
@@ -676,6 +676,26 @@
ao2_unlock(unid);
}
+static int apply_endpoint_acl(pjsip_rx_data *rdata, struct ast_sip_endpoint *endpoint);
+static int apply_endpoint_contact_acl(pjsip_rx_data *rdata, struct ast_sip_endpoint *endpoint);
+
+static void apply_acls(pjsip_rx_data *rdata)
+{
+ struct ast_sip_endpoint *endpoint;
+
+ /* Is the endpoint allowed with the source or contact address? */
+ endpoint = rdata->endpt_info.mod_data[endpoint_mod.id];
+ if (endpoint != artificial_endpoint
+ && (apply_endpoint_acl(rdata, endpoint)
+ || apply_endpoint_contact_acl(rdata, endpoint))) {
+ ast_debug(1, "Endpoint '%s' not allowed by ACL\n",
+ ast_sorcery_object_get_id(endpoint));
+
+ /* Replace the rdata endpoint with the artificial endpoint. */
+ ao2_replace(rdata->endpt_info.mod_data[endpoint_mod.id], artificial_endpoint);
+ }
+}
+
static pj_bool_t endpoint_lookup(pjsip_rx_data *rdata)
{
struct ast_sip_endpoint *endpoint;
@@ -695,6 +715,7 @@
ao2_unlink(unidentified_requests, unid);
ao2_ref(unid, -1);
}
+ apply_acls(rdata);
return PJ_FALSE;
}
@@ -759,6 +780,8 @@
ast_sip_report_invalid_endpoint(name, rdata);
}
}
+
+ apply_acls(rdata);
return PJ_FALSE;
}
@@ -842,16 +865,11 @@
ast_assert(endpoint != NULL);
- if (endpoint!=artificial_endpoint) {
- if (apply_endpoint_acl(rdata, endpoint) || apply_endpoint_contact_acl(rdata, endpoint)) {
- if (!is_ack) {
- pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 403, NULL, NULL, NULL);
- }
- return PJ_TRUE;
- }
+ if (is_ack) {
+ return PJ_FALSE;
}
- if (!is_ack && ast_sip_requires_authentication(endpoint, rdata)) {
+ if (ast_sip_requires_authentication(endpoint, rdata)) {
pjsip_tx_data *tdata;
struct unidentified_request *unid;
@@ -888,6 +906,10 @@
return PJ_TRUE;
}
pjsip_tx_data_dec_ref(tdata);
+ } else if (endpoint == artificial_endpoint) {
+ /* Uh. Oh. The artificial endpoint couldn't challenge so block the request. */
+ pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 500, NULL, NULL, NULL);
+ return PJ_TRUE;
}
return PJ_FALSE;

View file

@ -1,89 +0,0 @@
From 7d964e4b2e689f0dd7353e96d1782e92f59e9be6 Mon Sep 17 00:00:00 2001
From: Sean Bright <sean.bright@gmail.com>
Date: Thu, 16 Aug 2018 11:45:53 -0400
Subject: [PATCH] AST-2018-009: Fix crash processing websocket HTTP Upgrade
requests
The HTTP request processing in res_http_websocket allocates additional
space on the stack for various headers received during an Upgrade request.
An attacker could send a specially crafted request that causes this code
to overflow the stack, resulting in a crash.
* No longer allocate memory from the stack in a loop to parse the header
values. NOTE: There is a slight API change when using the passed in
strings as is. We now require the passed in strings to no longer have
leading or trailing whitespace. This isn't a problem as the only callers
have already done this before passing the strings to the affected
function.
ASTERISK-28013 #close
Change-Id: Ia564825a8a95e085fd17e658cb777fe1afa8091a
---
res/res_http_websocket.c | 25 ++++++++++++++-----------
1 file changed, 14 insertions(+), 11 deletions(-)
diff --git a/res/res_http_websocket.c b/res/res_http_websocket.c
index aaaba7d..e7ce830 100644
--- a/res/res_http_websocket.c
+++ b/res/res_http_websocket.c
@@ -758,7 +758,8 @@ static void websocket_bad_request(struct ast_tcptls_session_instance *ser)
int AST_OPTIONAL_API_NAME(ast_websocket_uri_cb)(struct ast_tcptls_session_instance *ser, const struct ast_http_uri *urih, const char *uri, enum ast_http_method method, struct ast_variable *get_vars, struct ast_variable *headers)
{
struct ast_variable *v;
- char *upgrade = NULL, *key = NULL, *key1 = NULL, *key2 = NULL, *protos = NULL, *requested_protocols = NULL, *protocol = NULL;
+ const char *upgrade = NULL, *key = NULL, *key1 = NULL, *key2 = NULL, *protos = NULL;
+ char *requested_protocols = NULL, *protocol = NULL;
int version = 0, flags = 1;
struct ast_websocket_protocol *protocol_handler = NULL;
struct ast_websocket *session;
@@ -777,16 +778,15 @@ int AST_OPTIONAL_API_NAME(ast_websocket_uri_cb)(struct ast_tcptls_session_instan
/* Get the minimum headers required to satisfy our needs */
for (v = headers; v; v = v->next) {
if (!strcasecmp(v->name, "Upgrade")) {
- upgrade = ast_strip(ast_strdupa(v->value));
+ upgrade = v->value;
} else if (!strcasecmp(v->name, "Sec-WebSocket-Key")) {
- key = ast_strip(ast_strdupa(v->value));
+ key = v->value;
} else if (!strcasecmp(v->name, "Sec-WebSocket-Key1")) {
- key1 = ast_strip(ast_strdupa(v->value));
+ key1 = v->value;
} else if (!strcasecmp(v->name, "Sec-WebSocket-Key2")) {
- key2 = ast_strip(ast_strdupa(v->value));
+ key2 = v->value;
} else if (!strcasecmp(v->name, "Sec-WebSocket-Protocol")) {
- requested_protocols = ast_strip(ast_strdupa(v->value));
- protos = ast_strdupa(requested_protocols);
+ protos = v->value;
} else if (!strcasecmp(v->name, "Sec-WebSocket-Version")) {
if (sscanf(v->value, "%30d", &version) != 1) {
version = 0;
@@ -800,7 +800,7 @@ int AST_OPTIONAL_API_NAME(ast_websocket_uri_cb)(struct ast_tcptls_session_instan
ast_sockaddr_stringify(&ser->remote_address));
ast_http_error(ser, 426, "Upgrade Required", NULL);
return 0;
- } else if (ast_strlen_zero(requested_protocols)) {
+ } else if (ast_strlen_zero(protos)) {
/* If there's only a single protocol registered, and the
* client doesn't specify what protocol it's using, go ahead
* and accept the connection */
@@ -821,9 +821,12 @@ int AST_OPTIONAL_API_NAME(ast_websocket_uri_cb)(struct ast_tcptls_session_instan
return 0;
}
- /* Iterate through the requested protocols trying to find one that we have a handler for */
- while (!protocol_handler && (protocol = strsep(&requested_protocols, ","))) {
- protocol_handler = ao2_find(server->protocols, ast_strip(protocol), OBJ_KEY);
+ if (!protocol_handler && protos) {
+ requested_protocols = ast_strdupa(protos);
+ /* Iterate through the requested protocols trying to find one that we have a handler for */
+ while (!protocol_handler && (protocol = strsep(&requested_protocols, ","))) {
+ protocol_handler = ao2_find(server->protocols, ast_strip(protocol), OBJ_KEY);
+ }
}
/* If no protocol handler exists bump this back to the requester */
--
2.7.4