Merge pull request #244 from micmac1/asterisk_I-for-17.01

Asterisk part 1 for 17.01

libs/pjproject/Makefile

@@ -9,12 +9,12 @@
 include $(TOPDIR)/rules.mk
 
 PKG_NAME:=pjproject
-PKG_VERSION:=2.4.5
+PKG_VERSION:=2.7.1
 PKG_RELEASE:=1
 
 PKG_SOURCE:=pjproject-$(PKG_VERSION).tar.bz2
 PKG_SOURCE_URL:=http://www.pjsip.org/release/$(PKG_VERSION)/
-PKG_MD5SUM:=f58b3485977b3a700256203a554b3869
+PKG_MD5SUM:=99a64110fa5c2debff40e0e8d4676380
 PKG_INSTALL:=1
 PKG_FIXUP:=autoreconf
 
@@ -31,7 +31,7 @@ define Package/pjproject/Default
   CATEGORY:=Libraries
   SUBMENU:=Telephony
   URL:=http://www.pjsip.org/
-  DEPENDS:=+libuuid +libstdcpp +libpthread
+  DEPENDS:=+libopenssl +libuuid +libstdcpp +libpthread
 endef
 
 define Package/pjproject/install/lib
@@ -54,46 +54,47 @@ $(call Package/pjproject/install/lib,$$(1),$2)
 endef
 
 CONFIGURE_ARGS += \
-	--enable-shared \
-	--disable-floating-point \
-	--enable-g711-codec \
-	--disable-l16-codec \
-	--disable-g722-codec \
-	--disable-g7221-codec \
-	--disable-gsm-codec \
-	--disable-ilbc-coder \
-	--disable-ipp \
-	--disable-ssl \
-	--disable-oss \
-	--disable-sound \
-	--with-external-srtp="$(STAGING_DIR)/usr" \
-	--without-external-gsm \
-	--disable-small-filter \
-	--disable-large-filter \
-	--disable-speex-aec \
-	--disable-g711-codec \
-	--disable-l16-codec \
-	--disable-gsm-codec \
-	--disable-g722-codec \
-	--disable-g7221-codec \
-	--disable-speex-codec \
-	--disable-ilbc-codec \
-	--disable-resample-dll \
-	--disable-sdl \
+	$(if $(CONFIG_SOFT_FLOAT),--disable-floating-point) \
+	--disable-bcg729 \
+	--disable-ext-sound \
 	--disable-ffmpeg \
-	--disable-v4l2
+	--disable-g711-codec \
+	--disable-g722-codec \
+	--disable-g7221-codec \
+	--disable-gsm-codec \
+	--disable-ilbc-codec \
+	--disable-ipp \
+	--disable-l16-codec \
+	--disable-libwebrtc \
+	--disable-libyuv \
+	--disable-opencore-amr \
+	--disable-openh264 \
+	--disable-opus \
+	--disable-oss \
+	--disable-resample \
+	--disable-sdl \
+	--disable-silk \
+	--disable-sound \
+	--disable-speex-aec \
+	--disable-speex-codec \
+	--disable-v4l2 \
+	--disable-video \
+	--enable-shared \
+	--with-external-srtp="$(STAGING_DIR)/usr" \
+	--with-ssl="$(STAGING_DIR)/usr" \
+	--without-external-gsm \
+	--without-external-pa \
+	--without-external-webrtc
 
-TARGET_LDFLAGS+=-lc $(LIBGCC) -lm
-TARGET_CFLAGS+=$(EXTRA_CFLAGS) $(TARGET_CPPFLAGS) $(EXTRA_CPPFLAGS)
+TARGET_CFLAGS+=$(TARGET_CPPFLAGS)
 
 define Build/Compile
 	$(MAKE) $(PKG_JOBS) -C $(PKG_BUILD_DIR)
 endef
 
 PJPROJECT_LIBS = \
-	libpj libpjlib-util libpjmedia-audiodev libpjmedia-codec \
-	libpjmedia-videodev libpjmedia libpjnath libpjsip-simple \
-	libpjsip-ua libpjsip libpjsua libpjsua2 libresample
+	libpj libpjlib-util libpjmedia libpjnath libpjsip-simple \
+	libpjsip-ua libpjsip libpjsua libpjsua2
 
 define Build/InstallDev
 	$(INSTALL_DIR) $(1)/usr/{include,lib}
@@ -102,16 +103,16 @@ define Build/InstallDev
 
 	$(foreach m,$(PJPROJECT_LIBS),$(CP) $(PKG_INSTALL_DIR)/usr/lib/$(m)* $(1)/usr/lib/;)
 	$(INSTALL_DIR) $(1)/usr/lib/pkgconfig
+	$(SED) 's|$(TARGET_CFLAGS)||g' $(PKG_INSTALL_DIR)/usr/lib/pkgconfig/libpjproject.pc
 	$(CP) $(PKG_INSTALL_DIR)/usr/lib/pkgconfig/libpjproject.pc $(1)/usr/lib/pkgconfig/
 endef
 
 $(eval $(call PJSIPpackage,libpj,libpj,+librt))
 $(eval $(call PJSIPpackage,libpjlib-util,libpjlib-util,+libpj +librt))
-$(eval $(call PJSIPpackage,libpjmedia,libpjmedia*,+libpj +libpjlib-util +libpjnath +libresample +librt +libspeex +libsrtp))
+$(eval $(call PJSIPpackage,libpjmedia,libpjmedia*,+libpj +libpjlib-util +libpjnath +librt +libsrtp))
 $(eval $(call PJSIPpackage,libpjnath,libpjnath,+libpj +libpjlib-util +librt))
-$(eval $(call PJSIPpackage,libpjsip-simple,libpjsip-simple,+libpj +libpjlib-util +libpjsip +libresample +librt +libspeex +libsrtp))
-$(eval $(call PJSIPpackage,libpjsip-ua,libpjsip-ua,+libpj +libpjlib-util +libpjmedia +libpjsip-simple +libpjsip +libresample +librt +libspeex +libsrtp))
-$(eval $(call PJSIPpackage,libpjsip,libpjsip,+libpj +libpjlib-util +libresample +librt +libspeex +libsrtp))
-$(eval $(call PJSIPpackage,libpjsua,libpjsua,+libpj +libpjlib-util +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +libresample +librt +libspeex +libsrtp))
-$(eval $(call PJSIPpackage,libpjsua2,libpjsua2,+libpj +libpjlib-util +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +libresample +librt +libspeex +libsrtp +libpjsua))
-$(eval $(call PJSIPpackage,libresample,libresample,))
+$(eval $(call PJSIPpackage,libpjsip-simple,libpjsip-simple,+libpj +libpjlib-util +libpjsip +librt))
+$(eval $(call PJSIPpackage,libpjsip-ua,libpjsip-ua,+libpj +libpjlib-util +libpjmedia +libpjsip-simple +libpjsip +librt))
+$(eval $(call PJSIPpackage,libpjsip,libpjsip,+libpj +libpjlib-util +librt +libsrtp))
+$(eval $(call PJSIPpackage,libpjsua,libpjsua,+libpj +libpjlib-util +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +librt))
+$(eval $(call PJSIPpackage,libpjsua2,libpjsua2,+libpj +libpjlib-util +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +librt +libpjsua))

libs/pjproject/patches/120-non-gnu-pthreads.patch

@@ -1,7 +1,5 @@
-Index: pjproject-2.4/pjlib/src/pj/os_core_unix.c
-===================================================================
---- pjproject-2.4.orig/pjlib/src/pj/os_core_unix.c
-+++ pjproject-2.4/pjlib/src/pj/os_core_unix.c
+--- pjproject-2.6/pjlib/src/pj/os_core_unix.c	2016-04-13 08:24:48.000000000 +0200
++++ pjproject-new/pjlib/src/pj/os_core_unix.c	2017-05-08 09:51:49.980905420 +0200
 @@ -1123,7 +1123,7 @@ static pj_status_t init_mutex(pj_mutex_t
  	return PJ_RETURN_OS_ERROR(rc);
  
@@ -9,7 +7,7 @@ Index: pjproject-2.4/pjlib/src/pj/os_core_unix.c
 -#if (defined(PJ_LINUX) && PJ_LINUX!=0) || \
 +#if (defined(PJ_LINUX) && PJ_LINUX!=0 && defined(__GLIBC__)) || \
      defined(PJ_HAS_PTHREAD_MUTEXATTR_SETTYPE)
- 	rc = pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_FAST_NP);
+ 	rc = pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_NORMAL);
  #elif (defined(PJ_RTEMS) && PJ_RTEMS!=0) || \
 @@ -1133,7 +1133,7 @@ static pj_status_t init_mutex(pj_mutex_t
  	rc = pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_NORMAL);
@@ -18,49 +16,5 @@ Index: pjproject-2.4/pjlib/src/pj/os_core_unix.c
 -#if (defined(PJ_LINUX) && PJ_LINUX!=0) || \
 +#if (defined(PJ_LINUX) && PJ_LINUX!=0 && defined(__GLIBC__)) || \
       defined(PJ_HAS_PTHREAD_MUTEXATTR_SETTYPE)
- 	rc = pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE_NP);
+ 	rc = pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE);
  #elif (defined(PJ_RTEMS) && PJ_RTEMS!=0) || \
-Index: pjproject-2.4/pjsip-apps/src/samples/siprtp.c
-===================================================================
---- pjproject-2.4.orig/pjsip-apps/src/samples/siprtp.c
-+++ pjproject-2.4/pjsip-apps/src/samples/siprtp.c
-@@ -1134,7 +1134,7 @@ static void boost_priority(void)
- 		    PJ_RETURN_OS_ERROR(rc));
- 	return;
-     }
--    tp.__sched_priority = max_prio;
-+    tp.sched_priority = max_prio;
- 
-     rc = sched_setscheduler(0, POLICY, &tp);
-     if (rc != 0) {
-@@ -1143,7 +1143,7 @@ static void boost_priority(void)
-     }
- 
-     PJ_LOG(4, (THIS_FILE, "New process policy=%d, priority=%d",
--	      policy, tp.__sched_priority));
-+	      policy, tp.sched_priority));
- 
-     /*
-      * Adjust thread scheduling algorithm and priority
-@@ -1156,10 +1156,10 @@ static void boost_priority(void)
-     }
- 
-     PJ_LOG(4, (THIS_FILE, "Old thread policy=%d, priority=%d",
--	      policy, tp.__sched_priority));
-+	      policy, tp.sched_priority));
- 
-     policy = POLICY;
--    tp.__sched_priority = max_prio;
-+    tp.sched_priority = max_prio;
- 
-     rc = pthread_setschedparam(pthread_self(), policy, &tp);
-     if (rc != 0) {
-@@ -1169,7 +1169,7 @@ static void boost_priority(void)
-     }
- 
-     PJ_LOG(4, (THIS_FILE, "New thread policy=%d, priority=%d",
--	      policy, tp.__sched_priority));
-+	      policy, tp.sched_priority));
- }
- 
- #else

libs/pjproject/patches/150-config_site.patch

@@ -0,0 +1,95 @@
+--- /dev/null
++++ b/pjlib/include/pj/config_site.h
+@@ -0,0 +1,92 @@
++/*
++ * Asterisk config_site.h
++ */
++
++#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
++#define PJ_IOQUEUE_MAX_HANDLES	(5000)
++#else
++#define PJ_IOQUEUE_MAX_HANDLES	(FD_SETSIZE)
++#endif
++#define PJ_IOQUEUE_HAS_SAFE_UNREG 1
++#define PJ_IOQUEUE_MAX_EVENTS_IN_SINGLE_POLL (16)
++
++#define PJ_SCANNER_USE_BITWISE	0
++#define PJ_OS_HAS_CHECK_STACK	0
++
++#ifndef PJ_LOG_MAX_LEVEL
++#define PJ_LOG_MAX_LEVEL		6
++#endif
++
++#define PJ_ENABLE_EXTRA_CHECK	1
++#define PJSIP_MAX_TSX_COUNT		((64*1024)-1)
++#define PJSIP_MAX_DIALOG_COUNT	((64*1024)-1)
++#define PJSIP_UDP_SO_SNDBUF_SIZE	(512*1024)
++#define PJSIP_UDP_SO_RCVBUF_SIZE	(512*1024)
++#define PJ_DEBUG			0
++#define PJSIP_SAFE_MODULE		0
++#define PJ_HAS_STRICMP_ALNUM		0
++
++/*
++ * Do not ever enable PJ_HASH_USE_OWN_TOLOWER because the algorithm is
++ * inconsistently used when calculating the hash value and doesn't
++ * convert the same characters as pj_tolower()/tolower().  Thus you
++ * can get different hash values if the string hashed has certain
++ * characters in it.  (ASCII '@', '[', '\\', ']', '^', and '_')
++ */
++#undef PJ_HASH_USE_OWN_TOLOWER
++
++/*
++  It is imperative that PJSIP_UNESCAPE_IN_PLACE remain 0 or undefined.
++  Enabling it will result in SEGFAULTS when URIs containing escape sequences are encountered.
++*/
++#undef PJSIP_UNESCAPE_IN_PLACE
++#define PJSIP_MAX_PKT_LEN			6000
++
++#undef PJ_TODO
++#define PJ_TODO(x)
++
++/* Defaults too low for WebRTC */
++#define PJ_ICE_MAX_CAND 32
++#define PJ_ICE_MAX_CHECKS (PJ_ICE_MAX_CAND * PJ_ICE_MAX_CAND)
++
++/* Increase limits to allow more formats */
++#define	PJMEDIA_MAX_SDP_FMT   64
++#define	PJMEDIA_MAX_SDP_BANDW   4
++#define	PJMEDIA_MAX_SDP_ATTR   (PJMEDIA_MAX_SDP_FMT*2 + 4)
++#define	PJMEDIA_MAX_SDP_MEDIA   16

net/asterisk-11.x/Makefile

@@ -10,7 +10,7 @@ include $(TOPDIR)/rules.mk
 
 PKG_NAME:=asterisk11
 PKG_VERSION:=11.22.0
-PKG_RELEASE:=2
+PKG_RELEASE:=3
 
 PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
 PKG_SOURCE_URL:=http://downloads.asterisk.org/pub/telephony/asterisk/releases/
@@ -146,6 +146,20 @@ $(foreach m,$(AST_EMB_MODULES),$(call Package/asterisk11/install/module,$(1),$(m
 	$(INSTALL_BIN) ./files/asterisk.init $(1)/etc/init.d/asterisk
 endef
 
+define Package/$(PKG_NAME)/postinst
+#!/bin/sh
+if [ -z "$${IPKG_INSTROOT}" ]; then
+  echo
+  echo "o-------------------------------------------------------------------o"
+  echo "| Asterisk 11 WARNING                                               |"
+  echo "o-------------------------------------------------------------------o"
+  echo "| Asterisk 11 is end-of-life. You should upgrade to Asterisk 13.    |"
+  echo "o-------------------------------------------------------------=^_^=-o"
+  echo
+fi
+exit 0
+endef
+
 define Package/asterisk11-sounds
 $(call Package/asterisk11/Default)
   TITLE:=Sounds support

net/asterisk-11.x/patches/054-AST-2016-007.patch

@@ -0,0 +1,117 @@
+From a503e4879cab7e35069e5481e0864b64b55e223d Mon Sep 17 00:00:00 2001
+From: Corey Farrell <git@cfware.com>
+Date: Mon, 8 Aug 2016 08:47:12 -0400
+Subject: [PATCH] Prevent leak of dialog RTP/SRTP instances.
+
+In some scenarios dialog_initialize_rtp can be called multiple times on
+the same dialog.  This can cause RTP instances to be leaked along with
+multiple file descriptors for each instance.
+
+ASTERISK-26272 #close
+
+Change-Id: Id716c2b87762d890c062b42538524a95067018a8
+---
+ channels/chan_sip.c | 61 ++++++++++++++++++++++++++++++++++-------------------
+ 1 file changed, 39 insertions(+), 22 deletions(-)
+
+diff --git a/channels/chan_sip.c b/channels/chan_sip.c
+index 9eaed58..2c29c9e 100644
+--- a/channels/chan_sip.c
++++ b/channels/chan_sip.c
+@@ -5697,6 +5697,38 @@ static void copy_socket_data(struct sip_socket *to_sock, const struct sip_socket
+ 	*to_sock = *from_sock;
+ }
+ 
++/*! Cleanup the RTP and SRTP portions of a dialog
++ *
++ * \note This procedure excludes vsrtp as it is initialized differently.
++ */
++static void dialog_clean_rtp(struct sip_pvt *p)
++{
++	if (p->rtp) {
++		ast_rtp_instance_destroy(p->rtp);
++		p->rtp = NULL;
++	}
++
++	if (p->vrtp) {
++		ast_rtp_instance_destroy(p->vrtp);
++		p->vrtp = NULL;
++	}
++
++	if (p->trtp) {
++		ast_rtp_instance_destroy(p->trtp);
++		p->trtp = NULL;
++	}
++
++	if (p->srtp) {
++		sip_srtp_destroy(p->srtp);
++		p->srtp = NULL;
++	}
++
++	if (p->tsrtp) {
++		sip_srtp_destroy(p->tsrtp);
++		p->tsrtp = NULL;
++	}
++}
++
+ /*! \brief Initialize DTLS-SRTP support on an RTP instance */
+ static int dialog_initialize_dtls_srtp(const struct sip_pvt *dialog, struct ast_rtp_instance *rtp, struct sip_srtp **srtp)
+ {
+@@ -5744,6 +5776,9 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog)
+ 		return 0;
+ 	}
+ 
++	/* Make sure previous RTP instances/FD's do not leak */
++	dialog_clean_rtp(dialog);
++
+ 	ast_sockaddr_copy(&bindaddr_tmp, &bindaddr);
+ 	if (!(dialog->rtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) {
+ 		return -1;
+@@ -6408,18 +6443,10 @@ static void sip_pvt_dtor(void *vdoomed)
+ 		ast_free(p->notify);
+ 		p->notify = NULL;
+ 	}
+-	if (p->rtp) {
+-		ast_rtp_instance_destroy(p->rtp);
+-		p->rtp = NULL;
+-	}
+-	if (p->vrtp) {
+-		ast_rtp_instance_destroy(p->vrtp);
+-		p->vrtp = NULL;
+-	}
+-	if (p->trtp) {
+-		ast_rtp_instance_destroy(p->trtp);
+-		p->trtp = NULL;
+-	}
++
++	/* Free RTP and SRTP instances */
++	dialog_clean_rtp(p);
++
+ 	if (p->udptl) {
+ 		ast_udptl_destroy(p->udptl);
+ 		p->udptl = NULL;
+@@ -6455,21 +6482,11 @@ static void sip_pvt_dtor(void *vdoomed)
+ 
+ 	destroy_msg_headers(p);
+ 
+-	if (p->srtp) {
+-		sip_srtp_destroy(p->srtp);
+-		p->srtp = NULL;
+-	}
+-
+ 	if (p->vsrtp) {
+ 		sip_srtp_destroy(p->vsrtp);
+ 		p->vsrtp = NULL;
+ 	}
+ 
+-	if (p->tsrtp) {
+-		sip_srtp_destroy(p->tsrtp);
+-		p->tsrtp = NULL;
+-	}
+-
+ 	if (p->directmediaacl) {
+ 		p->directmediaacl = ast_free_acl_list(p->directmediaacl);
+ 	}
+-- 
+2.5.5
+

net/asterisk-11.x/patches/055-AST-2016-009-11.diff

@@ -0,0 +1,27 @@
+diff --git a/channels/chan_sip.c b/channels/chan_sip.c
+index 556db57..9c74acb 100644
+--- a/channels/chan_sip.c
++++ b/channels/chan_sip.c
+@@ -8132,8 +8132,6 @@ static const char *__get_header(const struct sip_request *req, const char *name,
+ 	 * one afterwards.  If you shouldn't do it, what absolute idiot decided it was
+ 	 * a good idea to say you can do it, and if you can do it, why in the hell would.
+ 	 * you say you shouldn't.
+-	 * Anyways, pedanticsipchecking controls whether we allow spaces before ':',
+-	 * and we always allow spaces after that for compatibility.
+ 	 */
+ 	const char *sname = find_alias(name, NULL);
+ 	int x, len = strlen(name), slen = (sname ? 1 : 0);
+@@ -8146,10 +8144,10 @@ static const char *__get_header(const struct sip_request *req, const char *name,
+ 		if (match || smatch) {
+ 			/* skip name */
+ 			const char *r = header + (match ? len : slen );
+-			if (sip_cfg.pedanticsipchecking) {
+-				r = ast_skip_blanks(r);
++			/* HCOLON has optional SP/HTAB; skip past those */
++			while (*r == ' ' || *r == '\t') {
++				++r;
+ 			}
+-
+ 			if (*r == ':') {
+ 				*start = x+1;
+ 				return ast_skip_blanks(r+1);

net/asterisk-11.x/patches/056-AST-2017-005-11.diff

@@ -0,0 +1,195 @@
+From dc4c130439f053592b86f0b35c1fb219a0dc6587 Mon Sep 17 00:00:00 2001
+From: Joshua Colp <jcolp@digium.com>
+Date: Mon, 22 May 2017 15:36:38 +0000
+Subject: [PATCH] res_rtp_asterisk: Only learn a new source in learn state.
+
+This change moves the logic which learns a new source address
+for RTP so it only occurs in the learning state. The learning
+state is entered on initial allocation of RTP or if we are
+told that the remote address for the media has changed. While
+in the learning state if we continue to receive media from
+the original source we restart the learning process. It is
+only once we receive a sufficient number of RTP packets from
+the new source that we will switch to it. Once this is done
+the closed state is entered where all packets that do not
+originate from the expected source are dropped.
+
+The learning process has also been improved to take into
+account the time between received packets so a flood of them
+while in the learning state does not cause media to be switched.
+
+Finally RTCP now drops packets which are not for the learned
+SSRC if strict RTP is enabled.
+
+ASTERISK-27013
+
+Change-Id: I56a96e993700906355e79bc880ad9d4ad3ab129c
+---
+
+diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
+index 4cdc750..4881171 100644
+--- a/res/res_rtp_asterisk.c
++++ b/res/res_rtp_asterisk.c
+@@ -201,6 +201,7 @@
+ struct rtp_learning_info {
+ 	int max_seq;	/*!< The highest sequence number received */
+ 	int packets;	/*!< The number of remaining packets before the source is accepted */
++	struct timeval received; /*!< The time of the last received packet */
+ };
+ 
+ #ifdef HAVE_OPENSSL_SRTP
+@@ -286,7 +287,6 @@
+ 	 * but these are in place to keep learning mode sequence values sealed from their normal counterparts.
+ 	 */
+ 	struct rtp_learning_info rtp_source_learn;	/* Learning mode track for the expected RTP source */
+-	struct rtp_learning_info alt_source_learn;	/* Learning mode tracking for a new RTP source after one has been chosen */
+ 
+ 	struct rtp_red *red;
+ 
+@@ -2357,6 +2357,7 @@
+ {
+ 	info->max_seq = seq - 1;
+ 	info->packets = learning_min_sequential;
++	memset(&info->received, 0, sizeof(info->received));
+ }
+ 
+ /*!
+@@ -2371,6 +2372,13 @@
+  */
+ static int rtp_learning_rtp_seq_update(struct rtp_learning_info *info, uint16_t seq)
+ {
++	if (!ast_tvzero(info->received) && ast_tvdiff_ms(ast_tvnow(), info->received) < 5) {
++		/* During the probation period the minimum amount of media we'll accept is
++		 * 10ms so give a reasonable 5ms buffer just in case we get it sporadically.
++		 */
++		return 1;
++	}
++
+ 	if (seq == info->max_seq + 1) {
+ 		/* packet is in sequence */
+ 		info->packets--;
+@@ -2379,6 +2387,7 @@
+ 		info->packets = learning_min_sequential - 1;
+ 	}
+ 	info->max_seq = seq;
++	info->received = ast_tvnow();
+ 
+ 	return (info->packets == 0);
+ }
+@@ -2540,7 +2549,6 @@
+ 	rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_LEARN : STRICT_RTP_OPEN);
+ 	if (strictrtp) {
+ 		rtp_learning_seq_init(&rtp->rtp_source_learn, (uint16_t)rtp->seqno);
+-		rtp_learning_seq_init(&rtp->alt_source_learn, (uint16_t)rtp->seqno);
+ 	}
+ 
+ 	/* Create a new socket for us to listen on and use */
+@@ -3910,16 +3918,6 @@
+ 
+ 	packetwords = res / 4;
+ 
+-	if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT)) {
+-		/* Send to whoever sent to us */
+-		if (ast_sockaddr_cmp(&rtp->rtcp->them, &addr)) {
+-			ast_sockaddr_copy(&rtp->rtcp->them, &addr);
+-			if (rtpdebug)
+-				ast_debug(0, "RTCP NAT: Got RTCP from other end. Now sending to address %s\n",
+-					  ast_sockaddr_stringify(&rtp->rtcp->them));
+-		}
+-	}
+-
+ 	ast_debug(1, "Got RTCP report of %d bytes\n", res);
+ 
+ 	while (position < packetwords) {
+@@ -3939,6 +3937,24 @@
+ 			if (rtpdebug)
+ 				ast_debug(1, "RTCP Read too short\n");
+ 			return &ast_null_frame;
++		}
++
++		if ((rtp->strict_rtp_state != STRICT_RTP_OPEN) && (ntohl(rtcpheader[i + 1]) != rtp->themssrc)) {
++			/* Skip over this RTCP record as it does not contain the correct SSRC */
++			position += (length + 1);
++			ast_debug(1, "%p -- Received RTCP report from %s, dropping due to strict RTP protection. Received SSRC '%u' but expected '%u'\n",
++				rtp, ast_sockaddr_stringify(&addr), ntohl(rtcpheader[i + 1]), rtp->themssrc);
++			continue;
++		}
++
++		if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT)) {
++			/* Send to whoever sent to us */
++			if (ast_sockaddr_cmp(&rtp->rtcp->them, &addr)) {
++				ast_sockaddr_copy(&rtp->rtcp->them, &addr);
++				if (rtpdebug)
++					ast_debug(0, "RTCP NAT: Got RTCP from other end. Now sending to address %s\n",
++						ast_sockaddr_stringify(&rtp->rtcp->them));
++			}
+ 		}
+ 
+ 		if (rtcp_debug_test_addr(&addr)) {
+@@ -4330,24 +4346,11 @@
+ 
+ 	/* If strict RTP protection is enabled see if we need to learn the remote address or if we need to drop the packet */
+ 	if (rtp->strict_rtp_state == STRICT_RTP_LEARN) {
+-		ast_debug(1, "%p -- Probation learning mode pass with source address %s\n", rtp, ast_sockaddr_stringify(&addr));
+-		/* For now, we always copy the address. */
+-		ast_sockaddr_copy(&rtp->strict_rtp_address, &addr);
+-
+-		/* Send the rtp and the seqno from header to rtp_learning_rtp_seq_update to see whether we can exit or not*/
+-		if (rtp_learning_rtp_seq_update(&rtp->rtp_source_learn, seqno)) {
+-			ast_debug(1, "%p -- Probation at seq %d with %d to go; discarding frame\n",
+-				rtp, rtp->rtp_source_learn.max_seq, rtp->rtp_source_learn.packets);
+-			return &ast_null_frame;
+-		}
+-
+-		ast_verb(4, "%p -- Probation passed - setting RTP source address to %s\n", rtp, ast_sockaddr_stringify(&addr));
+-		rtp->strict_rtp_state = STRICT_RTP_CLOSED;
+-	}
+-	if (rtp->strict_rtp_state == STRICT_RTP_CLOSED) {
+ 		if (!ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
+-			/* Always reset the alternate learning source */
+-			rtp_learning_seq_init(&rtp->alt_source_learn, seqno);
++			/* We are learning a new address but have received traffic from the existing address,
++			 * accept it but reset the current learning for the new source so it only takes over
++			 * once sufficient traffic has been received. */
++			rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
+ 		} else {
+ 			/* Hmm, not the strict address. Perhaps we're getting audio from the alternate? */
+ 			if (!ast_sockaddr_cmp(&rtp->alt_rtp_address, &addr)) {
+@@ -4359,15 +4362,21 @@
+ 				 * it, that means we've stopped getting RTP from the original source and we should
+ 				 * switch to it.
+ 				 */
+-				if (rtp_learning_rtp_seq_update(&rtp->alt_source_learn, seqno)) {
++				if (rtp_learning_rtp_seq_update(&rtp->rtp_source_learn, seqno)) {
+ 					ast_debug(1, "%p -- Received RTP packet from %s, dropping due to strict RTP protection. Will switch to it in %d packets\n",
+-							rtp, ast_sockaddr_stringify(&addr), rtp->alt_source_learn.packets);
++							rtp, ast_sockaddr_stringify(&addr), rtp->rtp_source_learn.packets);
+ 					return &ast_null_frame;
+ 				}
+-				ast_verb(4, "%p -- Switching RTP source address to %s\n", rtp, ast_sockaddr_stringify(&addr));
+ 				ast_sockaddr_copy(&rtp->strict_rtp_address, &addr);
+ 			}
++
++			ast_verb(4, "%p -- Probation passed - setting RTP source address to %s\n", rtp, ast_sockaddr_stringify(&addr));
++			rtp->strict_rtp_state = STRICT_RTP_CLOSED;
+ 		}
++	} else if (rtp->strict_rtp_state == STRICT_RTP_CLOSED && ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
++		ast_debug(1, "%p -- Received RTP packet from %s, dropping due to strict RTP protection.\n",
++			rtp, ast_sockaddr_stringify(&addr));
++		return &ast_null_frame;
+ 	}
+ 
+ 	/* If symmetric RTP is enabled see if the remote side is not what we expected and change where we are sending audio */
+@@ -4762,7 +4771,11 @@
+ 
+ 	rtp->rxseqno = 0;
+ 
+-	if (strictrtp && rtp->strict_rtp_state != STRICT_RTP_OPEN) {
++	if (strictrtp && rtp->strict_rtp_state != STRICT_RTP_OPEN && !ast_sockaddr_isnull(addr) &&
++		ast_sockaddr_cmp(addr, &rtp->strict_rtp_address)) {
++		/* We only need to learn a new strict source address if we've been told the source is
++		 * changing to something different.
++		 */
+ 		rtp->strict_rtp_state = STRICT_RTP_LEARN;
+ 		rtp_learning_seq_init(&rtp->rtp_source_learn, rtp->seqno);
+ 	}

net/asterisk-11.x/patches/057-AST-2017-006-11.diff

@@ -0,0 +1,397 @@
+From 31676ce058596b57e10fbf83ff1817ca7907c3b1 Mon Sep 17 00:00:00 2001
+From: Corey Farrell <git@cfware.com>
+Date: Sat, 01 Jul 2017 20:24:27 -0400
+Subject: [PATCH] AST-2017-006: Fix app_minivm application MinivmNotify command injection
+
+An admin can configure app_minivm with an externnotify program to be run
+when a voicemail is received.  The app_minivm application MinivmNotify
+uses ast_safe_system() for this purpose which is vulnerable to command
+injection since the Caller-ID name and number values given to externnotify
+can come from an external untrusted source.
+
+* Add ast_safe_execvp() function.  This gives modules the ability to run
+external commands with greater safety compared to ast_safe_system().
+Specifically when some parameters are filled by untrusted sources the new
+function does not allow malicious input to break argument encoding.  This
+may be of particular concern where CALLERID(name) or CALLERID(num) may be
+used as a parameter to a script run by ast_safe_system() which could
+potentially allow arbitrary command execution.
+
+* Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp()
+instead of ast_safe_system() to avoid command injection.
+
+* Document code injection potential from untrusted data sources for other
+shell commands that are under user control.
+
+ASTERISK-27103
+
+Change-Id: I7552472247a84cde24e1358aaf64af160107aef1
+---
+
+diff --git a/README-SERIOUSLY.bestpractices.txt b/README-SERIOUSLY.bestpractices.txt
+index 281d0d3..d63f1df 100644
+--- a/README-SERIOUSLY.bestpractices.txt
++++ b/README-SERIOUSLY.bestpractices.txt
+@@ -94,6 +94,13 @@
+ ways in which you can mitigate this impact: stricter pattern matching, or using
+ the FILTER() dialplan function.
+ 
++The CALLERID(num) and CALLERID(name) values are other commonly used values that
++are sources of data potentially supplied by outside sources.  If you use these
++values as parameters to the System(), MixMonitor(), or Monitor() applications
++or the SHELL() dialplan function, you can allow injection of arbitrary operating
++system command execution.  The FILTER() dialplan function is available to remove
++dangerous characters from untrusted strings to block the command injection.
++
+ Strict Pattern Matching
+ -----------------------
+ 
+diff --git a/apps/app_minivm.c b/apps/app_minivm.c
+index ecdf9c6..8edc132 100644
+--- a/apps/app_minivm.c
++++ b/apps/app_minivm.c
+@@ -1741,21 +1741,35 @@
+ /*! \brief Run external notification for voicemail message */
+ static void run_externnotify(struct ast_channel *chan, struct minivm_account *vmu)
+ {
+-	char arguments[BUFSIZ];
++	char fquser[AST_MAX_CONTEXT * 2];
++	char *argv[5] = { NULL };
++	struct ast_party_caller *caller;
++	char *cid;
++	int idx;
+ 
+-	if (ast_strlen_zero(vmu->externnotify) && ast_strlen_zero(global_externnotify))
++	if (ast_strlen_zero(vmu->externnotify) && ast_strlen_zero(global_externnotify)) {
+ 		return;
++	}
+ 
+-	snprintf(arguments, sizeof(arguments), "%s %s@%s %s %s&", 
+-		ast_strlen_zero(vmu->externnotify) ? global_externnotify : vmu->externnotify, 
+-		vmu->username, vmu->domain,
+-		(ast_channel_caller(chan)->id.name.valid && ast_channel_caller(chan)->id.name.str)
+-			? ast_channel_caller(chan)->id.name.str : "",
+-		(ast_channel_caller(chan)->id.number.valid && ast_channel_caller(chan)->id.number.str)
+-			? ast_channel_caller(chan)->id.number.str : "");
++	snprintf(fquser, sizeof(fquser), "%s@%s", vmu->username, vmu->domain);
+ 
+-	ast_debug(1, "Executing: %s\n", arguments);
+-	ast_safe_system(arguments);
++	caller = ast_channel_caller(chan);
++	idx = 0;
++	argv[idx++] = ast_strlen_zero(vmu->externnotify) ? global_externnotify : vmu->externnotify;
++	argv[idx++] = fquser;
++	cid = S_COR(caller->id.name.valid, caller->id.name.str, NULL);
++	if (cid) {
++		argv[idx++] = cid;
++	}
++	cid = S_COR(caller->id.number.valid, caller->id.number.str, NULL);
++	if (cid) {
++		argv[idx++] = cid;
++	}
++	argv[idx] = NULL;
++
++	ast_debug(1, "Executing: %s %s %s %s\n",
++		argv[0], argv[1], argv[2] ?: "", argv[3] ?: "");
++	ast_safe_execvp(1, argv[0], argv);
+ }
+ 
+ /*!\internal
+diff --git a/apps/app_mixmonitor.c b/apps/app_mixmonitor.c
+index 89a1d8c..96adb9a 100644
+--- a/apps/app_mixmonitor.c
++++ b/apps/app_mixmonitor.c
+@@ -127,6 +127,11 @@
+ 				<para>Will be executed when the recording is over.</para>
+ 				<para>Any strings matching <literal>^{X}</literal> will be unescaped to <variable>X</variable>.</para>
+ 				<para>All variables will be evaluated at the time MixMonitor is called.</para>
++				<warning><para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
++				or <variable>CALLERID(name)</variable> as part of the command parameters.  You
++				risk a command injection attack executing arbitrary commands if the untrusted
++				strings aren't filtered to remove dangerous characters.  See function
++				<variable>FILTER()</variable>.</para></warning>
+ 			</parameter>
+ 		</syntax>
+ 		<description>
+@@ -143,6 +148,11 @@
+ 					<para>Will contain the filename used to record.</para>
+ 				</variable>
+ 			</variablelist>
++			<warning><para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
++			or <variable>CALLERID(name)</variable> as part of ANY of the application's
++			parameters.  You risk a command injection attack executing arbitrary commands
++			if the untrusted strings aren't filtered to remove dangerous characters.  See
++			function <variable>FILTER()</variable>.</para></warning>
+ 		</description>
+ 		<see-also>
+ 			<ref type="application">Monitor</ref>
+diff --git a/apps/app_system.c b/apps/app_system.c
+index 7fe453d..e868a07 100644
+--- a/apps/app_system.c
++++ b/apps/app_system.c
+@@ -48,6 +48,11 @@
+ 		<syntax>
+ 			<parameter name="command" required="true">
+ 				<para>Command to execute</para>
++				<warning><para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
++				or <variable>CALLERID(name)</variable> as part of the command parameters.  You
++				risk a command injection attack executing arbitrary commands if the untrusted
++				strings aren't filtered to remove dangerous characters.  See function
++				<variable>FILTER()</variable>.</para></warning>
+ 			</parameter>
+ 		</syntax>
+ 		<description>
+@@ -73,6 +78,11 @@
+ 		<syntax>
+ 			<parameter name="command" required="true">
+ 				<para>Command to execute</para>
++				<warning><para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
++				or <variable>CALLERID(name)</variable> as part of the command parameters.  You
++				risk a command injection attack executing arbitrary commands if the untrusted
++				strings aren't filtered to remove dangerous characters.  See function
++				<variable>FILTER()</variable>.</para></warning>
+ 			</parameter>
+ 		</syntax>
+ 		<description>
+diff --git a/configs/minivm.conf.sample b/configs/minivm.conf.sample
+index 55a39c8..3dcd59d 100644
+--- a/configs/minivm.conf.sample
++++ b/configs/minivm.conf.sample
+@@ -51,7 +51,7 @@
+ ; If you need to have an external program, i.e. /usr/bin/myapp called when a
+ ; voicemail is received by the server. The arguments are
+ ;
+-; 	<app> <username@domain> <callerid-number> <callerid-name>
++; 	<app> <username@domain> <callerid-name> <callerid-number>
+ ;
+ ;externnotify=/usr/bin/myapp
+ ; The character set for voicemail messages can be specified here
+diff --git a/funcs/func_shell.c b/funcs/func_shell.c
+index e403efc..79b7f99 100644
+--- a/funcs/func_shell.c
++++ b/funcs/func_shell.c
+@@ -84,6 +84,11 @@
+ 		<syntax>
+ 			<parameter name="command" required="true">
+ 				<para>The command that the shell should execute.</para>
++				<warning><para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
++				or <variable>CALLERID(name)</variable> as part of the command parameters.  You
++				risk a command injection attack executing arbitrary commands if the untrusted
++				strings aren't filtered to remove dangerous characters.  See function
++				<variable>FILTER()</variable>.</para></warning>
+ 			</parameter>
+ 		</syntax>
+ 		<description>
+diff --git a/include/asterisk/app.h b/include/asterisk/app.h
+index d10a0a6..8cdaea1 100644
+--- a/include/asterisk/app.h
++++ b/include/asterisk/app.h
+@@ -577,9 +577,34 @@
+ int ast_vm_test_create_user(const char *context, const char *mailbox);
+ #endif
+ 
+-/*! \brief Safely spawn an external program while closing file descriptors
+-	\note This replaces the \b system call in all Asterisk modules
+-*/
++/*!
++ * \brief Safely spawn an external program while closing file descriptors
++ *
++ * \note This replaces the \b execvp call in all Asterisk modules
++ *
++ * \param dualfork Non-zero to simulate running the program in the
++ * background by forking twice.  The option provides similar
++ * functionality to the '&' in the OS shell command "cmd &".  The
++ * option allows Asterisk to run a reaper loop to watch the first fork
++ * which immediately exits after spaning the second fork.  The actual
++ * program is run in the second fork.
++ * \param file execvp(file, argv) file parameter
++ * \param argv execvp(file, argv) argv parameter
++ */
++int ast_safe_execvp(int dualfork, const char *file, char *const argv[]);
++
++/*!
++ * \brief Safely spawn an OS shell command while closing file descriptors
++ *
++ * \note This replaces the \b system call in all Asterisk modules
++ *
++ * \param s - OS shell command string to execute.
++ *
++ * \warning Command injection can happen using this call if the passed
++ * in string is created using untrusted data from an external source.
++ * It is best not to use untrusted data.  However, the caller could
++ * filter out dangerous characters to avoid command injection.
++ */
+ int ast_safe_system(const char *s);
+ 
+ /*!
+diff --git a/main/asterisk.c b/main/asterisk.c
+index ce1d153..92256bd 100644
+--- a/main/asterisk.c
++++ b/main/asterisk.c
+@@ -1102,12 +1102,10 @@
+ 	ast_mutex_unlock(&safe_system_lock);
+ }
+ 
+-int ast_safe_system(const char *s)
++/*! \brief fork and perform other preparations for spawning applications */
++static pid_t safe_exec_prep(int dualfork)
+ {
+ 	pid_t pid;
+-	int res;
+-	struct rusage rusage;
+-	int status;
+ 
+ #if defined(HAVE_WORKING_FORK) || defined(HAVE_WORKING_VFORK)
+ 	ast_replace_sigchld();
+@@ -1129,35 +1127,102 @@
+ 		cap_free(cap);
+ #endif
+ #ifdef HAVE_WORKING_FORK
+-		if (ast_opt_high_priority)
++		if (ast_opt_high_priority) {
+ 			ast_set_priority(0);
++		}
+ 		/* Close file descriptors and launch system command */
+ 		ast_close_fds_above_n(STDERR_FILENO);
+ #endif
+-		execl("/bin/sh", "/bin/sh", "-c", s, (char *) NULL);
+-		_exit(1);
+-	} else if (pid > 0) {
++		if (dualfork) {
++#ifdef HAVE_WORKING_FORK
++			pid = fork();
++#else
++			pid = vfork();
++#endif
++			if (pid < 0) {
++				/* Second fork failed. */
++				/* No logger available. */
++				_exit(1);
++			}
++
++			if (pid > 0) {
++				/* This is the first fork, exit so the reaper finishes right away. */
++				_exit(0);
++			}
++
++			/* This is the second fork.  The first fork will exit immediately so
++			 * Asterisk doesn't have to wait for completion.
++			 * ast_safe_system("cmd &") would run in the background, but the '&'
++			 * cannot be added with ast_safe_execvp, so we have to double fork.
++			 */
++		}
++	}
++
++	if (pid < 0) {
++		ast_log(LOG_WARNING, "Fork failed: %s\n", strerror(errno));
++	}
++#else
++	ast_log(LOG_WARNING, "Fork failed: %s\n", strerror(ENOTSUP));
++	pid = -1;
++#endif
++
++	return pid;
++}
++
++/*! \brief wait for spawned application to complete and unreplace sigchld */
++static int safe_exec_wait(pid_t pid)
++{
++	int res = -1;
++
++#if defined(HAVE_WORKING_FORK) || defined(HAVE_WORKING_VFORK)
++	if (pid > 0) {
+ 		for (;;) {
++			struct rusage rusage;
++			int status;
++
+ 			res = wait4(pid, &status, 0, &rusage);
+ 			if (res > -1) {
+ 				res = WIFEXITED(status) ? WEXITSTATUS(status) : -1;
+ 				break;
+-			} else if (errno != EINTR)
++			}
++			if (errno != EINTR) {
+ 				break;
++			}
+ 		}
+-	} else {
+-		ast_log(LOG_WARNING, "Fork failed: %s\n", strerror(errno));
+-		res = -1;
+ 	}
+ 
+ 	ast_unreplace_sigchld();
+-#else /* !defined(HAVE_WORKING_FORK) && !defined(HAVE_WORKING_VFORK) */
+-	res = -1;
+ #endif
+ 
+ 	return res;
+ }
+ 
++int ast_safe_execvp(int dualfork, const char *file, char *const argv[])
++{
++	pid_t pid = safe_exec_prep(dualfork);
++
++	if (pid == 0) {
++		execvp(file, argv);
++		_exit(1);
++		/* noreturn from _exit */
++	}
++
++	return safe_exec_wait(pid);
++}
++
++int ast_safe_system(const char *s)
++{
++	pid_t pid = safe_exec_prep(0);
++
++	if (pid == 0) {
++		execl("/bin/sh", "/bin/sh", "-c", s, (char *) NULL);
++		_exit(1);
++		/* noreturn from _exit */
++	}
++
++	return safe_exec_wait(pid);
++}
++
+ /*!
+  * \brief enable or disable a logging level to a specified console
+  */
+diff --git a/res/res_monitor.c b/res/res_monitor.c
+index 76c43e1..12f478a 100644
+--- a/res/res_monitor.c
++++ b/res/res_monitor.c
+@@ -57,17 +57,17 @@
+ 		<syntax>
+ 			<parameter name="file_format" argsep=":">
+ 				<argument name="file_format" required="true">
+-					<para>optional, if not set, defaults to <literal>wav</literal></para>
++					<para>Optional.  If not set, defaults to <literal>wav</literal></para>
+ 				</argument>
+ 				<argument name="urlbase" />
+ 			</parameter>
+ 			<parameter name="fname_base">
+-				<para>if set, changes the filename used to the one specified.</para>
++				<para>If set, changes the filename used to the one specified.</para>
+ 			</parameter>
+ 			<parameter name="options">
+ 				<optionlist>
+ 					<option name="m">
+-						<para>when the recording ends mix the two leg files into one and
++						<para>When the recording ends mix the two leg files into one and
+ 						delete the two leg files. If the variable <variable>MONITOR_EXEC</variable>
+ 						is set, the application referenced in it will be executed instead of
+ 						soxmix/sox and the raw leg files will NOT be deleted automatically.
+@@ -78,6 +78,13 @@
+ 						will be passed on as additional arguments to <variable>MONITOR_EXEC</variable>.
+ 						Both <variable>MONITOR_EXEC</variable> and the Mix flag can be set from the
+ 						administrator interface.</para>
++						<warning><para>Do not use untrusted strings such as
++						<variable>CALLERID(num)</variable> or <variable>CALLERID(name)</variable>
++						as part of <variable>MONITOR_EXEC</variable> or
++						<variable>MONITOR_EXEC_ARGS</variable>.  You risk a command injection
++						attack executing arbitrary commands if the untrusted strings aren't
++						filtered to remove dangerous characters.  See function
++						<variable>FILTER()</variable>.</para></warning>
+ 					</option>
+ 					<option name="b">
+ 						<para>Don't begin recording unless a call is bridged to another channel.</para>

net/asterisk-11.x/patches/058-AST-2017-008-11.diff

@@ -0,0 +1,778 @@
+From fe2ba2f3ca60d33bc789c6ae8e03ee26dc1b637c Mon Sep 17 00:00:00 2001
+From: Richard Mudgett <rmudgett@digium.com>
+Date: Wed, 13 Sep 2017 12:07:42 -0500
+Subject: [PATCH] AST-2017-008: Improve RTP and RTCP packet processing.
+
+Validate RTCP packets before processing them.
+
+* Validate that the received packet is of a minimum length and apply the
+RFC3550 RTCP packet validation checks.
+
+* Fixed potentially reading garbage beyond the received RTCP record data.
+
+* Fixed rtp->themssrc only being set once when the remote could change
+the SSRC.  We would effectively stop handling the RTCP statistic records.
+
+* Fixed rtp->themssrc to not treat a zero value as special by adding
+rtp->themssrc_valid to indicate if rtp->themssrc is available.
+
+ASTERISK-27274
+
+Make strict RTP learning more flexible.
+
+Direct media can cause strict RTP to attempt to learn a remote address
+again before it has had a chance to learn the remote address the first
+time.  Because of the rapid relearn requests, strict RTP could latch onto
+the first remote address and fail to latch onto the direct media remote
+address.  As a result, you have one way audio until the call is placed on
+and off hold.
+
+The new algorithm learns remote addresses for a set time (1.5 seconds)
+before locking the remote address.  In addition, we must see a configured
+number of remote packets from the same address in a row before switching.
+
+* Fixed strict RTP learning from always accepting the first new address
+packet as the new stream.
+
+* Fixed strict RTP to initialize the expected sequence number with the
+last received sequence number instead of the last transmitted sequence
+number.
+
+* Fixed the predicted next sequence number calculation in
+rtp_learning_rtp_seq_update() to handle overflow.
+
+ASTERISK-27252
+
+Change-Id: Ia2d3aa6e0f22906c25971e74f10027d96525f31c
+---
+
+diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
+index 4881171..7393d57 100644
+--- a/res/res_rtp_asterisk.c
++++ b/res/res_rtp_asterisk.c
+@@ -115,7 +115,9 @@
+ 	STRICT_RTP_CLOSED,   /*! Drop all RTP packets not coming from source that was learned */
+ };
+ 
+-#define DEFAULT_STRICT_RTP STRICT_RTP_CLOSED
++#define STRICT_RTP_LEARN_TIMEOUT	1500	/*!< milliseconds */
++
++#define DEFAULT_STRICT_RTP -1	/*!< Enabled */
+ #define DEFAULT_ICESUPPORT 1
+ 
+ extern struct ast_srtp_res *res_srtp;
+@@ -199,9 +201,11 @@
+ 
+ /*! \brief RTP learning mode tracking information */
+ struct rtp_learning_info {
++	struct ast_sockaddr proposed_address;	/*!< Proposed remote address for strict RTP */
++	struct timeval start;	/*!< The time learning mode was started */
++	struct timeval received; /*!< The time of the last received packet */
+ 	int max_seq;	/*!< The highest sequence number received */
+ 	int packets;	/*!< The number of remaining packets before the source is accepted */
+-	struct timeval received; /*!< The time of the last received packet */
+ };
+ 
+ #ifdef HAVE_OPENSSL_SRTP
+@@ -223,7 +227,7 @@
+ 	unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
+ 	unsigned int ssrc;		/*!< Synchronization source, RFC 3550, page 10. */
+ 	unsigned int themssrc;		/*!< Their SSRC */
+-	unsigned int rxssrc;
++	unsigned int themssrc_valid;	/*!< True if their SSRC is available. */
+ 	unsigned int lastts;
+ 	unsigned int lastrxts;
+ 	unsigned int lastividtimestamp;
+@@ -1655,8 +1659,6 @@
+ #endif
+ };
+ 
+-static void rtp_learning_seq_init(struct rtp_learning_info *info, uint16_t seq);
+-
+ #ifdef HAVE_OPENSSL_SRTP
+ static void dtls_perform_handshake(struct ast_rtp_instance *instance, struct dtls_details *dtls, int rtcp)
+ {
+@@ -1685,6 +1687,8 @@
+ #endif
+ 
+ #ifdef USE_PJPROJECT
++static void rtp_learning_start(struct ast_rtp *rtp);
++
+ static void ast_rtp_on_ice_complete(pj_ice_sess *ice, pj_status_t status)
+ {
+ 	struct ast_rtp_instance *instance = ice->user_data;
+@@ -1721,8 +1725,8 @@
+ 		return;
+ 	}
+ 
+-	rtp->strict_rtp_state = STRICT_RTP_LEARN;
+-	rtp_learning_seq_init(&rtp->rtp_source_learn, (uint16_t)rtp->seqno);
++	ast_verb(4, "%p -- Strict RTP learning after ICE completion\n", rtp);
++	rtp_learning_start(rtp);
+ }
+ 
+ static void ast_rtp_on_ice_rx_data(pj_ice_sess *ice, unsigned comp_id, unsigned transport_id, void *pkt, pj_size_t size, const pj_sockaddr_t *src_addr, unsigned src_addr_len)
+@@ -2355,7 +2359,7 @@
+  */
+ static void rtp_learning_seq_init(struct rtp_learning_info *info, uint16_t seq)
+ {
+-	info->max_seq = seq - 1;
++	info->max_seq = seq;
+ 	info->packets = learning_min_sequential;
+ 	memset(&info->received, 0, sizeof(info->received));
+ }
+@@ -2372,14 +2376,17 @@
+  */
+ static int rtp_learning_rtp_seq_update(struct rtp_learning_info *info, uint16_t seq)
+ {
++	/*
++	 * During the learning mode the minimum amount of media we'll accept is
++	 * 10ms so give a reasonable 5ms buffer just in case we get it sporadically.
++	 */
+ 	if (!ast_tvzero(info->received) && ast_tvdiff_ms(ast_tvnow(), info->received) < 5) {
+-		/* During the probation period the minimum amount of media we'll accept is
+-		 * 10ms so give a reasonable 5ms buffer just in case we get it sporadically.
++		/*
++		 * Reject a flood of packets as acceptable for learning.
++		 * Reset the needed packets.
+ 		 */
+-		return 1;
+-	}
+-
+-	if (seq == info->max_seq + 1) {
++		info->packets = learning_min_sequential - 1;
++	} else if (seq == (uint16_t) (info->max_seq + 1)) {
+ 		/* packet is in sequence */
+ 		info->packets--;
+ 	} else {
+@@ -2389,7 +2396,23 @@
+ 	info->max_seq = seq;
+ 	info->received = ast_tvnow();
+ 
+-	return (info->packets == 0);
++	return info->packets;
++}
++
++/*!
++ * \brief Start the strictrtp learning mode.
++ *
++ * \param rtp RTP session description
++ *
++ * \return Nothing
++ */
++static void rtp_learning_start(struct ast_rtp *rtp)
++{
++	rtp->strict_rtp_state = STRICT_RTP_LEARN;
++	memset(&rtp->rtp_source_learn.proposed_address, 0,
++		sizeof(rtp->rtp_source_learn.proposed_address));
++	rtp->rtp_source_learn.start = ast_tvnow();
++	rtp_learning_seq_init(&rtp->rtp_source_learn, (uint16_t) rtp->lastrxseqno);
+ }
+ 
+ #ifdef USE_PJPROJECT
+@@ -2546,10 +2569,7 @@
+ 	/* Set default parameters on the newly created RTP structure */
+ 	rtp->ssrc = ast_random();
+ 	rtp->seqno = ast_random() & 0xffff;
+-	rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_LEARN : STRICT_RTP_OPEN);
+-	if (strictrtp) {
+-		rtp_learning_seq_init(&rtp->rtp_source_learn, (uint16_t)rtp->seqno);
+-	}
++	rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_CLOSED : STRICT_RTP_OPEN);
+ 
+ 	/* Create a new socket for us to listen on and use */
+ 	if ((rtp->s =
+@@ -3867,13 +3887,86 @@
+ 	return &rtp->f;
+ }
+ 
++static const char *rtcp_payload_type2str(unsigned int pt)
++{
++	const char *str;
++
++	switch (pt) {
++	case RTCP_PT_SR:
++		str = "Sender Report";
++		break;
++	case RTCP_PT_RR:
++		str = "Receiver Report";
++		break;
++	case RTCP_PT_FUR:
++		/* Full INTRA-frame Request / Fast Update Request */
++		str = "H.261 FUR";
++		break;
++	case RTCP_PT_SDES:
++		str = "Source Description";
++		break;
++	case RTCP_PT_BYE:
++		str = "BYE";
++		break;
++	default:
++		str = "Unknown";
++		break;
++	}
++	return str;
++}
++
++/*
++ * Unshifted RTCP header bit field masks
++ */
++#define RTCP_LENGTH_MASK			0xFFFF
++#define RTCP_PAYLOAD_TYPE_MASK		0xFF
++#define RTCP_REPORT_COUNT_MASK		0x1F
++#define RTCP_PADDING_MASK			0x01
++#define RTCP_VERSION_MASK			0x03
++
++/*
++ * RTCP header bit field shift offsets
++ */
++#define RTCP_LENGTH_SHIFT			0
++#define RTCP_PAYLOAD_TYPE_SHIFT		16
++#define RTCP_REPORT_COUNT_SHIFT		24
++#define RTCP_PADDING_SHIFT			29
++#define RTCP_VERSION_SHIFT			30
++
++#define RTCP_VERSION				2U
++#define RTCP_VERSION_SHIFTED		(RTCP_VERSION << RTCP_VERSION_SHIFT)
++#define RTCP_VERSION_MASK_SHIFTED	(RTCP_VERSION_MASK << RTCP_VERSION_SHIFT)
++
++/*
++ * RTCP first packet record validity header mask and value.
++ *
++ * RFC3550 intentionally defines the encoding of RTCP_PT_SR and RTCP_PT_RR
++ * such that they differ in the least significant bit.  Either of these two
++ * payload types MUST be the first RTCP packet record in a compound packet.
++ *
++ * RFC3550 checks the padding bit in the algorithm they use to check the
++ * RTCP packet for validity.  However, we aren't masking the padding bit
++ * to check since we don't know if it is a compound RTCP packet or not.
++ */
++#define RTCP_VALID_MASK (RTCP_VERSION_MASK_SHIFTED | (((RTCP_PAYLOAD_TYPE_MASK & ~0x1)) << RTCP_PAYLOAD_TYPE_SHIFT))
++#define RTCP_VALID_VALUE (RTCP_VERSION_SHIFTED | (RTCP_PT_SR << RTCP_PAYLOAD_TYPE_SHIFT))
++
++#define RTCP_SR_BLOCK_WORD_LENGTH 5
++#define RTCP_RR_BLOCK_WORD_LENGTH 6
++#define RTCP_HEADER_SSRC_LENGTH   2
++
+ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
+ {
+ 	struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ 	struct ast_sockaddr addr;
+ 	unsigned char rtcpdata[8192 + AST_FRIENDLY_OFFSET];
+ 	unsigned int *rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET);
+-	int res, packetwords, position = 0;
++	int res;
++	unsigned int packetwords;
++	unsigned int position;
++	unsigned int first_word;
++	/*! True if we have seen an acceptable SSRC to learn the remote RTCP address */
++	unsigned int ssrc_seen;
+ 	struct ast_frame *f = &ast_null_frame;
+ 
+ 	/* Read in RTCP data from the socket */
+@@ -3918,56 +4011,170 @@
+ 
+ 	packetwords = res / 4;
+ 
+-	ast_debug(1, "Got RTCP report of %d bytes\n", res);
++	ast_debug(1, "Got RTCP report of %d bytes from %s\n",
++		res, ast_sockaddr_stringify(&addr));
+ 
++	/*
++	 * Validate the RTCP packet according to an adapted and slightly
++	 * modified RFC3550 validation algorithm.
++	 */
++	if (packetwords < RTCP_HEADER_SSRC_LENGTH) {
++		ast_debug(1, "%p -- RTCP from %s: Frame size (%u words) is too short\n",
++			rtp, ast_sockaddr_stringify(&addr), packetwords);
++		return &ast_null_frame;
++	}
++	position = 0;
++	first_word = ntohl(rtcpheader[position]);
++	if ((first_word & RTCP_VALID_MASK) != RTCP_VALID_VALUE) {
++		ast_debug(1, "%p -- RTCP from %s: Failed first packet validity check\n",
++			rtp, ast_sockaddr_stringify(&addr));
++		return &ast_null_frame;
++	}
++	do {
++		position += ((first_word >> RTCP_LENGTH_SHIFT) & RTCP_LENGTH_MASK) + 1;
++		if (packetwords <= position) {
++			break;
++		}
++		first_word = ntohl(rtcpheader[position]);
++	} while ((first_word & RTCP_VERSION_MASK_SHIFTED) == RTCP_VERSION_SHIFTED);
++	if (position != packetwords) {
++		ast_debug(1, "%p -- RTCP from %s: Failed packet version or length check\n",
++			rtp, ast_sockaddr_stringify(&addr));
++		return &ast_null_frame;
++	}
++
++	/*
++	 * Note: RFC3605 points out that true NAT (vs NAPT) can cause RTCP
++	 * to have a different IP address and port than RTP.  Otherwise, when
++	 * strictrtp is enabled we could reject RTCP packets not coming from
++	 * the learned RTP IP address if it is available.
++	 */
++
++	/*
++	 * strictrtp safety needs SSRC to match before we use the
++	 * sender's address for symmetrical RTP to send our RTCP
++	 * reports.
++	 *
++	 * If strictrtp is not enabled then claim to have already seen
++	 * a matching SSRC so we'll accept this packet's address for
++	 * symmetrical RTP.
++	 */
++	ssrc_seen = rtp->strict_rtp_state == STRICT_RTP_OPEN;
++
++	position = 0;
+ 	while (position < packetwords) {
+-		int i, pt, rc;
+-		unsigned int length, dlsr, lsr, msw, lsw, comp;
++		unsigned int i;
++		unsigned int pt;
++		unsigned int rc;
++		unsigned int ssrc;
++		/*! True if the ssrc value we have is valid and not garbage because it doesn't exist. */
++		unsigned int ssrc_valid;
++		unsigned int length;
++		unsigned int min_length;
++		unsigned int dlsr, lsr, msw, lsw, comp;
+ 		struct timeval now;
+ 		double rttsec, reported_jitter, reported_normdev_jitter_current, normdevrtt_current, reported_lost, reported_normdev_lost_current;
+ 		uint64_t rtt = 0;
+ 
+ 		i = position;
+-		length = ntohl(rtcpheader[i]);
+-		pt = (length & 0xff0000) >> 16;
+-		rc = (length & 0x1f000000) >> 24;
+-		length &= 0xffff;
++		first_word = ntohl(rtcpheader[i]);
++		pt = (first_word >> RTCP_PAYLOAD_TYPE_SHIFT) & RTCP_PAYLOAD_TYPE_MASK;
++		rc = (first_word >> RTCP_REPORT_COUNT_SHIFT) & RTCP_REPORT_COUNT_MASK;
++		/* RFC3550 says 'length' is the number of words in the packet - 1 */
++		length = ((first_word >> RTCP_LENGTH_SHIFT) & RTCP_LENGTH_MASK) + 1;
+ 
+-		if ((i + length) > packetwords) {
+-			if (rtpdebug)
+-				ast_debug(1, "RTCP Read too short\n");
++		/* Check expected RTCP packet record length */
++		min_length = RTCP_HEADER_SSRC_LENGTH;
++		switch (pt) {
++		case RTCP_PT_SR:
++			min_length += RTCP_SR_BLOCK_WORD_LENGTH;
++			/* fall through */
++		case RTCP_PT_RR:
++			min_length += (rc * RTCP_RR_BLOCK_WORD_LENGTH);
++			break;
++		case RTCP_PT_FUR:
++			break;
++		case RTCP_PT_SDES:
++		case RTCP_PT_BYE:
++			/*
++			 * There may not be a SSRC/CSRC present.  The packet is
++			 * useless but still valid if it isn't present.
++			 *
++			 * We don't know what min_length should be so disable the check
++			 */
++			min_length = length;
++			break;
++		default:
++			ast_debug(1, "%p -- RTCP from %s: %u(%s) skipping record\n",
++				rtp, ast_sockaddr_stringify(&addr), pt, rtcp_payload_type2str(pt));
++			if (rtcp_debug_test_addr(&addr)) {
++				ast_verbose("\n");
++				ast_verbose("RTCP from %s: %u(%s) skipping record\n",
++					ast_sockaddr_stringify(&addr), pt, rtcp_payload_type2str(pt));
++			}
++			position += length;
++			continue;
++		}
++		if (length < min_length) {
++			ast_debug(1, "%p -- RTCP from %s: %u(%s) length field less than expected minimum.  Min:%u Got:%u\n",
++				rtp, ast_sockaddr_stringify(&addr), pt, rtcp_payload_type2str(pt),
++				min_length - 1, length - 1);
+ 			return &ast_null_frame;
+ 		}
+ 
+-		if ((rtp->strict_rtp_state != STRICT_RTP_OPEN) && (ntohl(rtcpheader[i + 1]) != rtp->themssrc)) {
+-			/* Skip over this RTCP record as it does not contain the correct SSRC */
+-			position += (length + 1);
+-			ast_debug(1, "%p -- Received RTCP report from %s, dropping due to strict RTP protection. Received SSRC '%u' but expected '%u'\n",
+-				rtp, ast_sockaddr_stringify(&addr), ntohl(rtcpheader[i + 1]), rtp->themssrc);
+-			continue;
+-		}
+-
+-		if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT)) {
+-			/* Send to whoever sent to us */
+-			if (ast_sockaddr_cmp(&rtp->rtcp->them, &addr)) {
+-				ast_sockaddr_copy(&rtp->rtcp->them, &addr);
+-				if (rtpdebug)
+-					ast_debug(0, "RTCP NAT: Got RTCP from other end. Now sending to address %s\n",
+-						ast_sockaddr_stringify(&rtp->rtcp->them));
+-			}
++		/* Get the RTCP record SSRC if defined for the record */
++		ssrc_valid = 1;
++		switch (pt) {
++		case RTCP_PT_SR:
++		case RTCP_PT_RR:
++		case RTCP_PT_FUR:
++			ssrc = ntohl(rtcpheader[i + 1]);
++			break;
++		case RTCP_PT_SDES:
++		case RTCP_PT_BYE:
++		default:
++			ssrc = 0;
++			ssrc_valid = 0;
++			break;
+ 		}
+ 
+ 		if (rtcp_debug_test_addr(&addr)) {
+-			ast_verbose("\n\nGot RTCP from %s\n",
+-				    ast_sockaddr_stringify(&addr));
+-			ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown");
+-			ast_verbose("Reception reports: %d\n", rc);
+-			ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]);
++			ast_verbose("\n");
++			ast_verbose("RTCP from %s\n", ast_sockaddr_stringify(&addr));
++			ast_verbose("PT: %u(%s)\n", pt, rtcp_payload_type2str(pt));
++			ast_verbose("Reception reports: %u\n", rc);
++			ast_verbose("SSRC of sender: %u\n", ssrc);
+ 		}
+ 
+-		i += 2; /* Advance past header and ssrc */
++		if (ssrc_valid && rtp->themssrc_valid) {
++			if (ssrc != rtp->themssrc) {
++				/*
++				 * Skip over this RTCP record as it does not contain the
++				 * correct SSRC.  We should not act upon RTCP records
++				 * for a different stream.
++				 */
++				position += length;
++				ast_debug(1, "%p -- RTCP from %s: Skipping record, received SSRC '%u' != expected '%u'\n",
++					rtp, ast_sockaddr_stringify(&addr), ssrc, rtp->themssrc);
++				continue;
++			}
++			ssrc_seen = 1;
++		}
++
++		if (ssrc_seen && ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT)) {
++			/* Send to whoever sent to us */
++			if (ast_sockaddr_cmp(&rtp->rtcp->them, &addr)) {
++				ast_sockaddr_copy(&rtp->rtcp->them, &addr);
++				if (rtpdebug) {
++					ast_debug(0, "RTCP NAT: Got RTCP from other end. Now sending to address %s\n",
++						ast_sockaddr_stringify(&addr));
++				}
++			}
++		}
++
++		i += RTCP_HEADER_SSRC_LENGTH; /* Advance past header and ssrc */
+ 		if (rc == 0 && pt == RTCP_PT_RR) {      /* We're receiving a receiver report with no reports, which is ok */
+-			position += (length + 1);
++			position += length;
+ 			continue;
+ 		}
+ 
+@@ -3983,7 +4190,7 @@
+ 				ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2]));
+ 				ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4]));
+ 			}
+-			i += 5;
++			i += RTCP_SR_BLOCK_WORD_LENGTH;
+ 			if (rc < 1)
+ 				break;
+ 			/* Intentional fall through */
+@@ -4153,21 +4360,18 @@
+ 		case RTCP_PT_SDES:
+ 			if (rtcp_debug_test_addr(&addr))
+ 				ast_verbose("Received an SDES from %s\n",
+-					    ast_sockaddr_stringify(&rtp->rtcp->them));
++					ast_sockaddr_stringify(&addr));
+ 			break;
+ 		case RTCP_PT_BYE:
+ 			if (rtcp_debug_test_addr(&addr))
+ 				ast_verbose("Received a BYE from %s\n",
+-					    ast_sockaddr_stringify(&rtp->rtcp->them));
++					ast_sockaddr_stringify(&addr));
+ 			break;
+ 		default:
+-			ast_debug(1, "Unknown RTCP packet (pt=%d) received from %s\n",
+-				  pt, ast_sockaddr_stringify(&rtp->rtcp->them));
+ 			break;
+ 		}
+-		position += (length + 1);
++		position += length;
+ 	}
+-
+ 	rtp->rtcp->rtcp_info = 1;
+ 
+ 	return f;
+@@ -4344,39 +4548,156 @@
+ 		return &ast_null_frame;
+ 	}
+ 
++	/* If the version is not what we expected by this point then just drop the packet */
++	if (version != 2) {
++		return &ast_null_frame;
++	}
++
+ 	/* If strict RTP protection is enabled see if we need to learn the remote address or if we need to drop the packet */
+-	if (rtp->strict_rtp_state == STRICT_RTP_LEARN) {
+-		if (!ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
+-			/* We are learning a new address but have received traffic from the existing address,
+-			 * accept it but reset the current learning for the new source so it only takes over
+-			 * once sufficient traffic has been received. */
+-			rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
++	switch (rtp->strict_rtp_state) {
++	case STRICT_RTP_LEARN:
++		/*
++		 * Scenario setup:
++		 * PartyA -- Ast1 -- Ast2 -- PartyB
++		 *
++		 * The learning timeout is necessary for Ast1 to handle the above
++		 * setup where PartyA calls PartyB and Ast2 initiates direct media
++		 * between Ast1 and PartyB.  Ast1 may lock onto the Ast2 stream and
++		 * never learn the PartyB stream when it starts.  The timeout makes
++		 * Ast1 stay in the learning state long enough to see and learn the
++		 * RTP stream from PartyB.
++		 *
++		 * To mitigate against attack, the learning state cannot switch
++		 * streams while there are competing streams.  The competing streams
++		 * interfere with each other's qualification.  Once we accept a
++		 * stream and reach the timeout, an attacker cannot interfere
++		 * anymore.
++		 *
++		 * Here are a few scenarios and each one assumes that the streams
++		 * are continuous:
++		 *
++		 * 1) We already have a known stream source address and the known
++		 * stream wants to change to a new source address.  An attacking
++		 * stream will block learning the new stream source.  After the
++		 * timeout we re-lock onto the original stream source address which
++		 * likely went away.  The result is one way audio.
++		 *
++		 * 2) We already have a known stream source address and the known
++		 * stream doesn't want to change source addresses.  An attacking
++		 * stream will not be able to replace the known stream.  After the
++		 * timeout we re-lock onto the known stream.  The call is not
++		 * affected.
++		 *
++		 * 3) We don't have a known stream source address.  This presumably
++		 * is the start of a call.  Competing streams will result in staying
++		 * in learning mode until a stream becomes the victor and we reach
++		 * the timeout.  We cannot exit learning if we have no known stream
++		 * to lock onto.  The result is one way audio until there is a victor.
++		 *
++		 * If we learn a stream source address before the timeout we will be
++		 * in scenario 1) or 2) when a competing stream starts.
++		 */
++		if (!ast_sockaddr_isnull(&rtp->strict_rtp_address)
++			&& STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(), rtp->rtp_source_learn.start)) {
++			ast_verb(4, "%p -- Strict RTP learning complete - Locking on source address %s\n",
++				rtp, ast_sockaddr_stringify(&rtp->strict_rtp_address));
++			rtp->strict_rtp_state = STRICT_RTP_CLOSED;
++
++			/*
++			 * Clear the alternate remote address after learning.
++			 *
++			 * We should not leave this address laying around.
++			 * It gets set only on a chan_sip reINVITE glare.
++			 * We don't want a stale address interfering with
++			 * the next learning time.
++			 */
++			ast_sockaddr_setnull(&rtp->alt_rtp_address);
+ 		} else {
+-			/* Hmm, not the strict address. Perhaps we're getting audio from the alternate? */
+-			if (!ast_sockaddr_cmp(&rtp->alt_rtp_address, &addr)) {
+-				/* ooh, we did! You're now the new expected address, son! */
+-				ast_sockaddr_copy(&rtp->strict_rtp_address,
+-						  &addr);
+-			} else {
+-				/* Start trying to learn from the new address. If we pass a probationary period with
+-				 * it, that means we've stopped getting RTP from the original source and we should
+-				 * switch to it.
++			if (!ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
++				/*
++				 * We are open to learning a new address but have received
++				 * traffic from the current address, accept it and reset
++				 * the learning counts for a new source.  When no more
++				 * current source packets arrive a new source can take over
++				 * once sufficient traffic is received.
+ 				 */
+-				if (rtp_learning_rtp_seq_update(&rtp->rtp_source_learn, seqno)) {
+-					ast_debug(1, "%p -- Received RTP packet from %s, dropping due to strict RTP protection. Will switch to it in %d packets\n",
+-							rtp, ast_sockaddr_stringify(&addr), rtp->rtp_source_learn.packets);
+-					return &ast_null_frame;
+-				}
+-				ast_sockaddr_copy(&rtp->strict_rtp_address, &addr);
++				rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
++				break;
+ 			}
+ 
+-			ast_verb(4, "%p -- Probation passed - setting RTP source address to %s\n", rtp, ast_sockaddr_stringify(&addr));
+-			rtp->strict_rtp_state = STRICT_RTP_CLOSED;
++			/*
++			 * We give preferential treatment to the requested remote address
++			 * (negotiated SDP address) where we are to send our RTP.  However,
++			 * the other end has no obligation to send from that address even
++			 * though it is practically a requirement when NAT is involved.
++			 */
++			if (!ast_sockaddr_cmp(&remote_address, &addr)) {
++				/* Accept the negotiated remote RTP stream as the source */
++				ast_verb(4, "%p -- Strict RTP switching to RTP remote address %s as source\n",
++					rtp, ast_sockaddr_stringify(&addr));
++				ast_sockaddr_copy(&rtp->strict_rtp_address, &addr);
++				rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
++				break;
++			}
++			/* Treat the alternate remote address as another negotiated SDP address. */
++			if (!ast_sockaddr_isnull(&rtp->alt_rtp_address)
++				&& !ast_sockaddr_cmp(&rtp->alt_rtp_address, &addr)) {
++				/* ooh, we did! You're now the new expected address, son! */
++				ast_verb(4, "%p -- Strict RTP switching to RTP alt remote address %s as source\n",
++					rtp, ast_sockaddr_stringify(&addr));
++				ast_sockaddr_copy(&rtp->strict_rtp_address, &addr);
++				rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
++				break;
++			}
++
++			/*
++			 * Trying to learn a new address.  If we pass a probationary period
++			 * with it, that means we've stopped getting RTP from the original
++			 * source and we should switch to it.
++			 */
++			if (!ast_sockaddr_cmp(&rtp->rtp_source_learn.proposed_address, &addr)) {
++				if (!rtp_learning_rtp_seq_update(&rtp->rtp_source_learn, seqno)) {
++					/* Accept the new RTP stream */
++					ast_verb(4, "%p -- Strict RTP switching source address to %s\n",
++						rtp, ast_sockaddr_stringify(&addr));
++					ast_sockaddr_copy(&rtp->strict_rtp_address, &addr);
++					rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
++					break;
++				}
++				/* Not ready to accept the RTP stream candidate */
++				ast_debug(1, "%p -- Received RTP packet from %s, dropping due to strict RTP protection. Will switch to it in %d packets.\n",
++					rtp, ast_sockaddr_stringify(&addr), rtp->rtp_source_learn.packets);
++			} else {
++				/*
++				 * This is either an attacking stream or
++				 * the start of the expected new stream.
++				 */
++				ast_sockaddr_copy(&rtp->rtp_source_learn.proposed_address, &addr);
++				rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
++				ast_debug(1, "%p -- Received RTP packet from %s, dropping due to strict RTP protection. Qualifying new stream.\n",
++					rtp, ast_sockaddr_stringify(&addr));
++			}
++			return &ast_null_frame;
+ 		}
+-	} else if (rtp->strict_rtp_state == STRICT_RTP_CLOSED && ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
++		/* Fall through */
++	case STRICT_RTP_CLOSED:
++		/*
++		 * We should not allow a stream address change if the SSRC matches
++		 * once strictrtp learning is closed.  Any kind of address change
++		 * like this should have happened while we were in the learning
++		 * state.  We do not want to allow the possibility of an attacker
++		 * interfering with the RTP stream after the learning period.
++		 * An attacker could manage to get an RTCP packet redirected to
++		 * them which can contain the SSRC value.
++		 */
++		if (!ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
++			break;
++		}
+ 		ast_debug(1, "%p -- Received RTP packet from %s, dropping due to strict RTP protection.\n",
+ 			rtp, ast_sockaddr_stringify(&addr));
+ 		return &ast_null_frame;
++	case STRICT_RTP_OPEN:
++		break;
+ 	}
+ 
+ 	/* If symmetric RTP is enabled see if the remote side is not what we expected and change where we are sending audio */
+@@ -4401,11 +4722,6 @@
+ 		return &ast_null_frame;
+ 	}
+ 
+-	/* If the version is not what we expected by this point then just drop the packet */
+-	if (version != 2) {
+-		return &ast_null_frame;
+-	}
+-
+ 	/* Pull out the various other fields we will need */
+ 	payloadtype = (seqno & 0x7f0000) >> 16;
+ 	padding = seqno & (1 << 29);
+@@ -4418,7 +4734,7 @@
+ 
+ 	AST_LIST_HEAD_INIT_NOLOCK(&frames);
+ 	/* Force a marker bit and change SSRC if the SSRC changes */
+-	if (rtp->rxssrc && rtp->rxssrc != ssrc) {
++	if (rtp->themssrc_valid && rtp->themssrc != ssrc) {
+ 		struct ast_frame *f, srcupdate = {
+ 			AST_FRAME_CONTROL,
+ 			.subclass.integer = AST_CONTROL_SRCCHANGE,
+@@ -4445,8 +4761,8 @@
+ 			rtp->rtcp->received_prior = 0;
+ 		}
+ 	}
+-
+-	rtp->rxssrc = ssrc;
++	rtp->themssrc = ssrc; /* Record their SSRC to put in future RR */
++	rtp->themssrc_valid = 1;
+ 
+ 	/* Remove any padding bytes that may be present */
+ 	if (padding) {
+@@ -4498,10 +4814,6 @@
+ 
+ 	prev_seqno = rtp->lastrxseqno;
+ 	rtp->lastrxseqno = seqno;
+-
+-	if (!rtp->themssrc) {
+-		rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
+-	}
+ 
+ 	if (rtp_debug_test_addr(&addr)) {
+ 		ast_verbose("Got  RTP packet from    %s (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6d)\n",
+@@ -4771,13 +5083,14 @@
+ 
+ 	rtp->rxseqno = 0;
+ 
+-	if (strictrtp && rtp->strict_rtp_state != STRICT_RTP_OPEN && !ast_sockaddr_isnull(addr) &&
+-		ast_sockaddr_cmp(addr, &rtp->strict_rtp_address)) {
++	if (strictrtp && rtp->strict_rtp_state != STRICT_RTP_OPEN
++		&& !ast_sockaddr_isnull(addr) && ast_sockaddr_cmp(addr, &rtp->strict_rtp_address)) {
+ 		/* We only need to learn a new strict source address if we've been told the source is
+ 		 * changing to something different.
+ 		 */
+-		rtp->strict_rtp_state = STRICT_RTP_LEARN;
+-		rtp_learning_seq_init(&rtp->rtp_source_learn, rtp->seqno);
++		ast_verb(4, "%p -- Strict RTP learning after remote address set to: %s\n",
++			rtp, ast_sockaddr_stringify(addr));
++		rtp_learning_start(rtp);
+ 	}
+ 
+ 	return;
+@@ -4805,7 +5118,23 @@
+ 	 */
+ 	ast_sockaddr_copy(&rtp->alt_rtp_address, addr);
+ 
+-	return;
++	if (strictrtp && rtp->strict_rtp_state != STRICT_RTP_OPEN
++		&& !ast_sockaddr_isnull(addr) && ast_sockaddr_cmp(addr, &rtp->strict_rtp_address)) {
++		/*
++		 * We only need to learn a new strict source address if we've been told the
++		 * source may be changing to something different.
++		 *
++		 * XXX NOTE: The alternate source address is only set because of a reINVITE
++		 * glare in chan_sip.  A reINVITE glare is supposed to be retried after a
++		 * backoff delay so it shouldn't be needed at all.  However, I found this
++		 * as the best description of why it was added:
++		 * http://lists.digium.com/pipermail/asterisk-dev/2009-May/038348.html
++		 * https://reviewboard.asterisk.org/r/252/
++		 */
++		ast_verb(4, "%p -- Strict RTP learning after alternate remote address set to: %s\n",
++			rtp, ast_sockaddr_stringify(addr));
++		rtp_learning_start(rtp);
++	}
+ }
+ 
+ /*! \brief Write t140 redundacy frame

net/asterisk-13.x/Config.in

@@ -0,0 +1,11 @@
+menu "Advanced configuration"
+	depends on PACKAGE_asterisk13
+
+config ASTERISK13_LOW_MEMORY
+	bool "Optimize Asterisk 13 for low memory usage"
+	default n
+	help
+	  Warning: this feature is known to cause problems with some modules.
+	  Disable it if you experience problems like segmentation faults.
+
+endmenu

net/asterisk-13.x/Makefile

@@ -9,12 +9,12 @@
 include $(TOPDIR)/rules.mk
 
 PKG_NAME:=asterisk13
-PKG_VERSION:=13.9.1
+PKG_VERSION:=13.18.5
 PKG_RELEASE:=1
 
 PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
-PKG_SOURCE_URL:=http://downloads.asterisk.org/pub/telephony/asterisk/releases/
-PKG_MD5SUM:=76c42992a79f41ec467ed20500e8b249
+PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases/
+PKG_MD5SUM:=4ad2a5ab1dd12cba5f37fca52961aa2a
 
 PKG_BUILD_DIR:=$(BUILD_DIR)/asterisk-$(PKG_VERSION)
 PKG_BUILD_DEPENDS:=libxml2/host
@@ -46,8 +46,12 @@ define Package/asterisk13/install/sbin
 endef
 
 define Package/asterisk13/install/sounds
-	$(INSTALL_DIR) $(1)/usr/lib/asterisk/sounds/
-	$(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/sounds/en/$(2) $(1)/usr/lib/asterisk/sounds/
+	$(INSTALL_DIR) $(1)/usr/share/asterisk/sounds/
+	$(CP) $(PKG_INSTALL_DIR)/usr/share/asterisk/sounds/en/$(2) $(1)/usr/share/asterisk/sounds/
+endef
+
+define Package/$(PKG_NAME)/config
+	source "$(SOURCE)/Config.in"
 endef
 
 define BuildAsterisk13Module
@@ -58,7 +62,7 @@ define BuildAsterisk13Module
   endef
 
   define Package/asterisk13-$(1)/conffiles
-$(foreach c,$(5),/etc/asterisk/$(c))
+$(subst $(space),$(newline),$(foreach c,$(5),/etc/asterisk/$(c)))
   endef
 
   define Package/asterisk13-$(1)/description
@@ -105,7 +109,11 @@ define Package/asterisk13/conffiles
 /etc/asterisk/acl.conf
 /etc/asterisk/cel.conf
 /etc/asterisk/ccss.conf
-/etc/asterisk/modules.conf
+/etc/asterisk/cli.conf
+/etc/asterisk/cli_permissions.conf
+/etc/asterisk/codecs.conf
+/etc/asterisk/dnsmgr.conf
+/etc/asterisk/dsp.conf
 /etc/asterisk/extconfig.conf
 /etc/asterisk/extensions.conf
 /etc/asterisk/features.conf
@@ -115,7 +123,7 @@ define Package/asterisk13/conffiles
 /etc/asterisk/manager.conf
 /etc/asterisk/modules.conf
 /etc/asterisk/res_config_sqlite3.conf
-/etc/asterisk/rtp.conf
+/etc/asterisk/stasis.conf
 /etc/asterisk/udptl.conf
 /etc/asterisk/users.conf
 /etc/default/asterisk
@@ -123,9 +131,10 @@ define Package/asterisk13/conffiles
 endef
 
 AST_CFG_FILES:= \
-	asterisk.conf acl.conf cel.conf ccss.conf extconfig.conf \
+	asterisk.conf acl.conf cel.conf ccss.conf cli.conf \
+	cli_permissions.conf codecs.conf dnsmgr.conf dsp.conf extconfig.conf \
 	extensions.conf features.conf http.conf indications.conf \
-	logger.conf manager.conf modules.conf udptl.conf \
+	logger.conf manager.conf modules.conf stasis.conf udptl.conf \
 	users.conf res_config_sqlite3.conf
 
 AST_EMB_MODULES:=\
@@ -140,7 +149,7 @@ $(call Package/asterisk13/install/sbin,$(1),safe_asterisk)
 $(call Package/asterisk13/install/sbin,$(1),astgenkey)
 $(foreach m,$(AST_CFG_FILES),$(call Package/asterisk13/install/conffile,$(1),$(m));)
 $(foreach m,$(AST_EMB_MODULES),$(call Package/asterisk13/install/module,$(1),$(m));)
-	$(INSTALL_DIR) $(1)/usr/lib/asterisk/sounds/
+	$(INSTALL_DIR) $(1)/usr/share/asterisk/sounds/
 	$(INSTALL_DIR) $(1)/etc/default
 	$(INSTALL_DATA) ./files/asterisk.default $(1)/etc/default/asterisk
 	$(INSTALL_DIR) $(1)/etc/init.d
@@ -158,12 +167,12 @@ This package provides the sound-files for Asterisk-13.
 endef
 
 define Package/asterisk13-sounds/install
-	$(INSTALL_DIR) $(1)/usr/lib/asterisk/sounds/
-	$(CP) $(PKG_INSTALL_DIR)/usr/lib/asterisk/sounds/en/* $(1)/usr/lib/asterisk/sounds/
-	rm -f $(1)/usr/lib/asterisk/sounds/vm-*
+	$(INSTALL_DIR) $(1)/usr/share/asterisk/sounds/
+	$(CP) $(PKG_INSTALL_DIR)/usr/share/asterisk/sounds/en/* $(1)/usr/share/asterisk/sounds/
+	rm -f $(1)/usr/share/asterisk/sounds/vm-*
 endef
 
-ifneq ($(SDK)$(CONFIG_PACKAGE_asterisk13-chan-dahdi),)
+ifneq ($(CONFIG_PACKAGE_asterisk13-chan-dahdi),)
   CONFIGURE_ARGS+= \
 	--with-dahdi="$(STAGING_DIR)/usr" \
 	--with-pri="$(STAGING_DIR)/usr" \
@@ -175,13 +184,12 @@ else
 	--without-tonezone
 endif
 
-TARGET_LDFLAGS+= \
-	$(if $(CONFIG_PACKAGE_$(PKG_NAME)-pbx-lua),-ldl -lcrypt)
-
-EXTRA_CFLAGS+=$(TARGET_CPPFLAGS)
-EXTRA_LDFLAGS+=$(TARGET_LDFLAGS) -Wl,-rpath-link,$(STAGING_DIR)/usr/lib
+# Pass CPPFLAGS in the CFLAGS as otherwise the build system will
+# ignore them.
+TARGET_CFLAGS+=$(TARGET_CPPFLAGS)
 
 CONFIGURE_ARGS+= \
+	$(if $(CONFIG_PACKAGE_$(PKG_NAME)-chan-alsa),--with-asound="$(STAGING_DIR)/usr",--without-asound) \
 	--without-execinfo \
 	--without-bluetooth \
 	--without-cap \
@@ -203,30 +211,57 @@ CONFIGURE_ARGS+= \
 	--without-osptk \
 	$(if $(CONFIG_PACKAGE_$(PKG_NAME)-pbx-lua),--with-lua="$(STAGING_DIR)/usr",--without-lua) \
 	$(if $(CONFIG_PACKAGE_$(PKG_NAME)-pgsql),--with-postgres="$(STAGING_DIR)/usr",--without-postgres) \
-	$(if $(CONFIG_PACKAGE_$(PKG_NAME)-pjsip),--with-pjproject,--without-pjproject) \
 	--with-popt="$(STAGING_DIR)/usr" \
 	--without-pwlib \
 	--without-radius \
-	--without-spandsp \
+	$(if $(CONFIG_PACKAGE_$(PKG_NAME)-res-fax-spandsp),--with-spandsp="$(STAGING_DIR)/usr",--without-spandsp) \
 	$(if $(CONFIG_PACKAGE_$(PKG_NAME)-res-xmpp),--with-iksemel="$(STAGING_DIR)/usr",--without-iksemel) \
 	--without-sdl \
 	--without-sqlite \
 	--with-sqlite3="$(STAGING_DIR)/usr" \
-	$(if $(CONFIG_PACKAGE_$(PKG_NAME)-res-srtp),--with-srtp="$(STAGING_DIR)/usr",--without-srtp) \
 	--without-suppserv \
 	--without-tds \
 	--without-termcap \
 	--without-tinfo \
-	--with-uuid="$(STAGING_DIR)/usr" \
 	--without-vorbis \
 	--without-vpb \
 	--with-z="$(STAGING_DIR)/usr" \
 	--with-sounds-cache="$(DL_DIR)" \
 	--enable-xmldoc
 
+ifeq ($(CONFIG_PACKAGE_$(PKG_NAME)-res-pjproject)$(CONFIG_PACKAGE_$(PKG_NAME)-res-srtp),)
+CONFIGURE_ARGS+= \
+	--without-srtp
+else
+CONFIGURE_ARGS+= \
+	--with-srtp="$(STAGING_DIR)/usr"
+endif
+
+ifeq ($(CONFIG_PACKAGE_$(PKG_NAME)-pjsip)$(CONFIG_PACKAGE_$(PKG_NAME)-res-pjproject)$(CONFIG_PACKAGE_$(PKG_NAME)-res-rtp-asterisk),)
+CONFIGURE_ARGS+= \
+	--without-pjproject
+else
+CONFIGURE_ARGS+= \
+	--with-pjproject="$(STAGING_DIR)/usr"
+endif
+
 CONFIGURE_VARS += \
 	ac_cv_path_ac_pt_CONFIG_LIBXML2=$(STAGING_DIR)/host/bin/xml2-config
 
+MAKE_FLAGS+= \
+	ASTDATADIR="/usr/share/asterisk" \
+	DESTDIR="$(PKG_INSTALL_DIR)"
+
+# show full gcc arguments instead of [CC] and [LD]
+MAKE_FLAGS+= \
+	NOISY_BUILD="yes"
+
+# don't let asterisk mess with build flags
+MAKE_FLAGS+= \
+	AST_FORTIFY_SOURCE="" \
+	DEBUG="" \
+	OPTIMIZE=""
+
 AST_MENUSELECT_OPTS = \
 	--without-newt \
 	--without-curses \
@@ -237,7 +272,7 @@ define Build/Configure
 	(cd $(PKG_BUILD_DIR); \
 		./bootstrap.sh; \
 	);
-	$(call Build/Configure/Default,,$(SITE_VARS))
+	$(call Build/Configure/Default)
 	(cd $(PKG_BUILD_DIR)/menuselect; \
 		./bootstrap.sh; \
 		./configure \
@@ -252,22 +287,20 @@ define Build/Compile
 	$(MAKE) -C "$(PKG_BUILD_DIR)/menuselect" \
 		CFLAGS="$(HOST_CFLAGS) -I$(STAGING_DIR)/host/include/libxml2" \
 		LDFLAGS="$(HOST_LDFLAGS) -lxml2"
-	$(MAKE) -C "$(PKG_BUILD_DIR)" \
-		include/asterisk/version.h \
-		include/asterisk/buildopts.h defaults.h \
-		makeopts.embed_rules
-	ASTCFLAGS="$(EXTRA_CFLAGS) -DLOW_MEMORY"
-	ASTLDFLAGS="$(EXTRA_LDFLAGS)"
-	$(MAKE) -C "$(PKG_BUILD_DIR)" \
-		ASTVARLIBDIR="/usr/lib/asterisk" \
-		ASTDATADIR="/usr/lib/asterisk" \
-		ASTKEYDIR="/usr/lib/asterisk" \
-		ASTDBDIR="/usr/lib/asterisk" \
-		NOISY_BUILD="yes" \
-		DEBUG="" \
-		OPTIMIZE="" \
-		DESTDIR="$(PKG_INSTALL_DIR)" \
-		all install samples
+	$(MAKE) -C "$(PKG_BUILD_DIR)" menuselect-tree
+	cd "$(PKG_BUILD_DIR)" && \
+		./menuselect/menuselect \
+			--disable BUILD_NATIVE \
+			$(if $(CONFIG_ASTERISK13_LOW_MEMORY),--enable LOW_MEMORY) \
+			menuselect.makeopts
+	# Hack:
+	# When changing anything in MENUSELECT_CFLAGS the file ".lastclean"
+	# gets deleted. E.g. when compiling on x86 for x86 "--disable
+	# BUILD_NATIVE" changes MENUSELECT_CFLAGS and the file gets removed.
+	# But that will result in a rebuild attempt of menuselect which will
+	# likely fail. Prevent that by recreating ".lastclean".
+	$(CP) "$(PKG_BUILD_DIR)/.cleancount" "$(PKG_BUILD_DIR)/.lastclean"
+	$(call Build/Compile/Default,all install samples)
 endef
 
 define Build/InstallDev
@@ -329,7 +362,7 @@ $(eval $(call BuildAsterisk13Module,cdr,Provides CDR,Call Detail Record,,cdr.con
 $(eval $(call BuildAsterisk13Module,cdr-csv,Provides CDR CSV,Call Detail Record with CSV support,,,cdr_csv,,))
 $(eval $(call BuildAsterisk13Module,cdr-sqlite3,Provides CDR SQLITE3,Call Detail Record with SQLITE3 support,libsqlite3,,cdr_sqlite3_custom,,))
 $(eval $(call BuildAsterisk13Module,chan-alsa,ALSA channel,the channel chan_alsa,+alsa-lib,alsa.conf,chan_alsa,,))
-$(eval $(call BuildAsterisk13Module,chan-dahdi,DAHDI channel,DAHDI channel support,+dahdi-tools-libtonezone +kmod-dahdi +libpri,chan_dahdi.conf,chan_dahdi,,))
+$(eval $(call BuildAsterisk13Module,chan-dahdi,DAHDI channel,DAHDI channel support,+dahdi-tools-libtonezone +kmod-dahdi +libpri @!aarch64,chan_dahdi.conf,chan_dahdi,,))
 $(eval $(call BuildAsterisk13Module,chan-iax2,IAX2 channel,IAX support,+asterisk13-res-timing-timerfd,iax.conf iaxprov.conf,chan_iax2,,))
 $(eval $(call BuildAsterisk13Module,chan-oss,OSS channel,the channel chan_oss,,oss.conf,chan_oss,,))
 $(eval $(call BuildAsterisk13Module,chan-sip,SIP channel,the channel chan_sip,+asterisk13-app-confbridge,sip.conf sip_notify.conf,chan_sip,,))
@@ -346,7 +379,7 @@ $(eval $(call BuildAsterisk13Module,codec-ilbc,linear to ILBC translation,transl
 $(eval $(call BuildAsterisk13Module,codec-lpc10,Linear to LPC10 translation,translate between signed linear and LPC10,,,codec_lpc10,,))
 $(eval $(call BuildAsterisk13Module,codec-resample,resample sLinear audio,resample sLinear audio,,,codec_resample,,))
 $(eval $(call BuildAsterisk13Module,codec-ulaw,Signed linear to ulaw translation,translation between signed linear and ulaw codecs,,,codec_ulaw,,))
-$(eval $(call BuildAsterisk13Module,curl,CURL,CURL support,+libcurl,,func_curl res_curl,,))
+$(eval $(call BuildAsterisk13Module,curl,CURL,CURL support,+libcurl,,func_curl res_config_curl res_curl,,))
 $(eval $(call BuildAsterisk13Module,format-g726,G.726,support for headerless G.726 16/24/32/40kbps data format,,,format_g726,,))
 $(eval $(call BuildAsterisk13Module,format-g729,G.729,support for raw headerless G729 data,,,format_g729,,))
 $(eval $(call BuildAsterisk13Module,format-gsm,GSM format,support for GSM format,,,format_gsm,,))
@@ -372,38 +405,44 @@ $(eval $(call BuildAsterisk13Module,func-groupcount,Group count,for counting num
 $(eval $(call BuildAsterisk13Module,func-math,Math functions,Math functions,,,func_math,))
 $(eval $(call BuildAsterisk13Module,func-module,Simple module check function,Simple module check function,,,func_module,))
 $(eval $(call BuildAsterisk13Module,func-presencestate,Hinted presence state,Gets or sets a presence state in the dialplan,,,func_presencestate,,))
+$(eval $(call BuildAsterisk13Module,func-periodic-hook,Periodic dialplan hooks,Execute a periodic dialplan hook into the audio of a call,+$(PKG_NAME)-app-chanspy +$(PKG_NAME)-func-cut +$(PKG_NAME)-func-groupcount +$(PKG_NAME)-func-uri,,func_periodic_hook,,))
 $(eval $(call BuildAsterisk13Module,func-realtime,realtime,the realtime dialplan function,,,func_realtime,,))
 $(eval $(call BuildAsterisk13Module,func-shell,Shell,support for shell execution,,,func_shell,,))
 $(eval $(call BuildAsterisk13Module,func-uri,URI encoding and decoding,Encodes and decodes URI-safe strings,,,func_uri,,))
 $(eval $(call BuildAsterisk13Module,func-vmcount,vmcount dialplan,a vmcount dialplan function,,,func_vmcount,,))
-$(eval $(call BuildAsterisk13Module,odbc,ODBC,ODBC support,+libpthread +libc +unixodbc,cdr_adaptive_odbc.conf cdr_odbc.conf cel_odbc.conf func_odbc.conf res_odbc.conf,cdr_adaptive_odbc cdr_odbc cel_odbc func_odbc res_config_odbc res_odbc,,))
-$(eval $(call BuildAsterisk13Module,pbx-ael,Asterisk Extension Logic,support for symbolic Asterisk Extension Logic,,extensions.ael,pbx_ael,,))
+$(eval $(call BuildAsterisk13Module,odbc,ODBC,ODBC support,+libpthread +libc +unixodbc,cdr_adaptive_odbc.conf cdr_odbc.conf cel_odbc.conf func_odbc.conf res_odbc.conf,cdr_adaptive_odbc cdr_odbc cel_odbc func_odbc res_config_odbc res_odbc res_odbc_transaction,,))
+$(eval $(call BuildAsterisk13Module,pbx-ael,Asterisk Extension Logic,support for symbolic Asterisk Extension Logic,+$(PKG_NAME)-res-ael-share,extensions.ael,pbx_ael,,))
 $(eval $(call BuildAsterisk13Module,pbx-dundi,Dundi,provides Dundi Lookup service for Asterisk,,dundi.conf,pbx_dundi,,))
 $(eval $(call BuildAsterisk13Module,pbx-realtime,Realtime Switch,realtime switch support,,,pbx_realtime,,))
 $(eval $(call BuildAsterisk13Module,pbx-spool,Call Spool,outgoing call spool support,,,pbx_spool,,))
-$(eval $(call BuildAsterisk13Module,pgsql,PostgreSQL,PostgreSQL support,+libpq,cel_pgsql.conf cdr_pgsql.conf res_pgsql.conf,cel_pgsql cdr_pgsql res_config_pgsql,,))
-$(eval $(call BuildAsterisk13Module,pjsip,pjsip channel,the channel pjsip,+asterisk13-res-sorcery +libpjsip +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsua +libpjsua2,pjsip.conf pjsip_notify.conf,func_pjsip_endpoint chan_pjsip res_pjsip_acl res_pjsip_authenticator_digest res_pjsip_caller_id res_pjsip_dialog_info_body_generator res_pjsip_diversion res_pjsip_dtmf_info res_pjsip_endpoint_identifier_anonymous res_pjsip_endpoint_identifier_ip res_pjsip_endpoint_identifier_user res_pjsip_exten_state res_pjsip_header_funcs res_pjsip_log_forwarder res_pjsip_logger res_pjsip_messaging res_pjsip_multihomed res_pjsip_mwi_body_generator res_pjsip_mwi res_pjsip_nat res_pjsip_notify res_pjsip_one_touch_record_info res_pjsip_outbound_authenticator_digest res_pjsip_outbound_publish res_pjsip_outbound_registration res_pjsip_path res_pjsip_pidf_body_generator res_pjsip_pidf_digium_body_supplement res_pjsip_pidf_eyebeam_body_supplement res_pjsip_publish_asterisk res_pjsip_pubsub res_pjsip_refer res_pjsip_registrar_expire res_pjsip_registrar res_pjsip_rfc3326 res_pjsip_sdp_rtp res_pjsip_send_to_voicemail res_pjsip_session res_pjsip res_pjsip_transport_websocket res_pjsip_t38 res_pjsip_xpidf_body_generator,,))
+$(eval $(call BuildAsterisk13Module,pgsql,PostgreSQL,PostgreSQL support,+libpq @!arc,cel_pgsql.conf cdr_pgsql.conf res_pgsql.conf,cel_pgsql cdr_pgsql res_config_pgsql,,))
+$(eval $(call BuildAsterisk13Module,pjsip,pjsip channel,the channel pjsip,+asterisk13-res-sorcery +asterisk13-res-pjproject +libpjsip +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsua +libpjsua2,pjsip.conf pjsip_notify.conf pjsip_wizard.conf,chan_pjsip func_pjsip_aor func_pjsip_contact func_pjsip_endpoint res_pjsip res_pjsip_acl res_pjsip_authenticator_digest res_pjsip_caller_id res_pjsip_config_wizard res_pjsip_dialog_info_body_generator res_pjsip_diversion res_pjsip_dlg_options res_pjsip_dtmf_info res_pjsip_empty_info res_pjsip_endpoint_identifier_anonymous res_pjsip_endpoint_identifier_ip res_pjsip_endpoint_identifier_user res_pjsip_exten_state res_pjsip_header_funcs res_pjsip_history res_pjsip_logger res_pjsip_messaging res_pjsip_mwi res_pjsip_mwi_body_generator res_pjsip_nat res_pjsip_notify res_pjsip_one_touch_record_info res_pjsip_outbound_authenticator_digest res_pjsip_outbound_publish res_pjsip_outbound_registration res_pjsip_path res_pjsip_pidf_body_generator res_pjsip_pidf_digium_body_supplement res_pjsip_pidf_eyebeam_body_supplement res_pjsip_publish_asterisk res_pjsip_pubsub res_pjsip_refer res_pjsip_registrar res_pjsip_registrar_expire res_pjsip_rfc3326 res_pjsip_sdp_rtp res_pjsip_send_to_voicemail res_pjsip_session res_pjsip_sips_contact res_pjsip_t38 res_pjsip_transport_management res_pjsip_transport_websocket res_pjsip_xpidf_body_generator,,))
 $(eval $(call BuildAsterisk13Module,res-adsi,Provide ADSI,Analog Display Services Interface capability,,,res_adsi,,))
 $(eval $(call BuildAsterisk13Module,res-ael-share,Shareable AEL code,support for shareable AEL code mainly between internal and external modules,,,res_ael_share,,))
-$(eval $(call BuildAsterisk13Module,res-agi,Asterisk Gateway Interface,Support for the Asterisk Gateway Interface extension,,,res_agi,,))
+$(eval $(call BuildAsterisk13Module,res-agi,Asterisk Gateway Interface,Support for the Asterisk Gateway Interface extension,+asterisk13-res-speech,,res_agi,,))
 $(eval $(call BuildAsterisk13Module,res-calendar,Calendaring API,Calendaring support (ICal and Google Calendar),,calendar.conf,res_calendar,,))
 $(eval $(call BuildAsterisk13Module,res-clioriginate,Calls via CLI,Originate calls via the CLI,,,res_clioriginate,,))
-$(eval $(call BuildAsterisk13Module,res-hep,HEPv3 API,,,,res_hep,,))
-$(eval $(call BuildAsterisk13Module,res-hep-pjsip,PJSIP HEPv3 Logger,,+asterisk13-res-hep +asterisk13-pjsip,,res_hep,,))
-$(eval $(call BuildAsterisk13Module,res-hep-rtcp,RTCP HEPv3 Logger,,+asterisk13-res-hep,,res_hep,,))
-$(eval $(call BuildAsterisk13Module,res-http-websocket,HTTP websocket support,,,,res_http_websocket,,))
-$(eval $(call BuildAsterisk13Module,res-monitor,Provide Monitor,Cryptographic Signature capability,,,res_monitor,,))
+$(eval $(call BuildAsterisk13Module,res-fax,FAX modules,Generic FAX resource for FAX technology resource modules,+asterisk13-res-timing-pthread,res_fax.conf,res_fax,,))
+$(eval $(call BuildAsterisk13Module,res-fax-spandsp,Spandsp T.38 and G.711,Spandsp T.38 and G.711 FAX Resource,+asterisk13-res-fax +libspandsp +libtiff,,res_fax_spandsp,,))
+$(eval $(call BuildAsterisk13Module,res-hep,HEPv3 API,Routines for integration with Homer using HEPv3,,hep.conf,res_hep,,))
+$(eval $(call BuildAsterisk13Module,res-hep-pjsip,PJSIP HEPv3 Logger,PJSIP logging with Homer,+asterisk13-res-hep +asterisk13-pjsip,,res_hep_pjsip,,))
+$(eval $(call BuildAsterisk13Module,res-hep-rtcp,RTCP HEPv3 Logger,RTCP logging with Homer,+asterisk13-res-hep,,res_hep_rtcp,,))
+$(eval $(call BuildAsterisk13Module,res-http-websocket,HTTP websocket support,WebSocket support for the Asterisk internal HTTP server,,,res_http_websocket,,))
+$(eval $(call BuildAsterisk13Module,res-monitor,PBX channel monitoring,call monitoring resource,+$(PKG_NAME)-func-periodic-hook,,res_monitor,,))
 $(eval $(call BuildAsterisk13Module,res-musiconhold,MOH,Music On Hold support,,musiconhold.conf,res_musiconhold,,))
-$(eval $(call BuildAsterisk13Module,res-parking,Phone Parking,Phone Parking application,,res_parking.conf,res_parking,,))
+$(eval $(call BuildAsterisk13Module,res-parking,Phone Parking,Phone Parking application,+$(PKG_NAME)-bridge-holding,res_parking.conf,res_parking,,))
 $(eval $(call BuildAsterisk13Module,res-phoneprov,Phone Provisioning,Phone provisioning application for the asterisk internal http server,,phoneprov.conf,res_phoneprov,,))
+$(eval $(call BuildAsterisk13Module,res-pjproject,Bridge PJPROJECT to Asterisk logging,,+libpj +libpjlib-util +libpjmedia +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +libpjsua +libpjsua2 +libsrtp,pjproject.conf,res_pjproject,,))
 $(eval $(call BuildAsterisk13Module,res-realtime,Realtime,Realtime Interface,,,res_realtime,,))
-$(eval $(call BuildAsterisk13Module,res-rtp-asterisk,RTP stack,,+libpjsip +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsua +libpjsua2,rtp.conf,res_rtp_asterisk,,))
-$(eval $(call BuildAsterisk13Module,res-rtp-multicast,RTP multicast engine,,,,res_rtp_multicast,,))
+$(eval $(call BuildAsterisk13Module,res-rtp-asterisk,RTP stack,Supports RTP and RTCP with Symmetric RTP support for NAT traversal,+libpjsip +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsua +libpjsua2,rtp.conf,res_rtp_asterisk,,))
+$(eval $(call BuildAsterisk13Module,res-rtp-multicast,RTP multicast engine,Multicast RTP Engine,,,res_rtp_multicast,,))
 $(eval $(call BuildAsterisk13Module,res-smdi,Provide SMDI,Simple Message Desk Interface capability,,smdi.conf,res_smdi,,))
-$(eval $(call BuildAsterisk13Module,res-sorcery,Sorcery data layer,,,,res_sorcery_astdb res_sorcery_config res_sorcery_memory res_sorcery_realtime,,))
+$(eval $(call BuildAsterisk13Module,res-sorcery,Sorcery data layer,Sorcery backend modules for data access intended for using realtime as backend,,sorcery.conf,res_sorcery_astdb res_sorcery_config res_sorcery_memory res_sorcery_realtime,,))
+$(eval $(call BuildAsterisk13Module,res-speech,Speech Recognition API,Support for the Asterisk Generic Speech Recognition API,,,res_speech,,))
 $(eval $(call BuildAsterisk13Module,res-srtp,SRTP Support,Secure RTP connection,+libsrtp,,res_srtp,,))
-$(eval $(call BuildAsterisk13Module,res-timing-dahdi,DAHDI Timing Interface,,+asterisk13-chan-dahdi,,res_timing_dahdi,,))
-$(eval $(call BuildAsterisk13Module,res-timing-pthread,pthread Timing Interface,,,,res_timing_pthread,,))
-$(eval $(call BuildAsterisk13Module,res-timing-timerfd,Timerfd Timing Interface,,,,res_timing_timerfd,,))
+$(eval $(call BuildAsterisk13Module,res-timing-dahdi,DAHDI Timing Interface,DAHDI timing interface,+asterisk13-chan-dahdi,,res_timing_dahdi,,))
+$(eval $(call BuildAsterisk13Module,res-timing-pthread,pthread Timing Interface,POSIX pthreads Timing Interface,,,res_timing_pthread,,))
+$(eval $(call BuildAsterisk13Module,res-timing-timerfd,Timerfd Timing Interface,Timing interface provided by Linux kernel,,,res_timing_timerfd,,))
+$(eval $(call BuildAsterisk13Module,res-xmpp,XMPP client and component module,reference module for interfacting Asterisk directly as a client or component with XMPP server,+libiksemel +libopenssl,xmpp.conf,res_xmpp,,))
 $(eval $(call BuildAsterisk13Module,voicemail,Voicemail,voicemail related modules,+asterisk13-res-adsi +asterisk13-res-smdi,voicemail.conf,app_voicemail,vm-*,))
 

net/asterisk-13.x/files/asterisk.init

@@ -14,8 +14,7 @@ start() {
        [ -d $DEST/var/run/asterisk ] || mkdir -p $DEST/var/run/asterisk
        [ -d $DEST/var/log/asterisk ] || mkdir -p $DEST/var/log/asterisk
        [ -d $DEST/var/spool/asterisk ] || mkdir -p $DEST/var/spool/asterisk
-       [ -d $DEST/var/lib ] || mkdir -p $DEST/var/lib
-       [ -h $DEST/var/lib/asterisk ] || ln -s /usr/lib/asterisk /var/lib/asterisk
+       [ -d $DEST/var/lib/asterisk ] || mkdir -p $DEST/var/lib/asterisk
        [ -d $DEST/var/lib/asterisk/keys ] || mkdir -p $DEST/var/lib/asterisk/keys
        [ -d $DEST/var/log/asterisk/cdr-csv ] || mkdir -p $DEST/var/log/asterisk/cdr-csv
 

net/asterisk-13.x/patches/001-disable-semaphores-check.patch

@@ -1,6 +1,6 @@
 --- a/configure.ac
 +++ b/configure.ac
-@@ -927,19 +927,6 @@ AC_LINK_IFELSE(
+@@ -963,19 +963,6 @@ AC_LINK_IFELSE(
    ]
  )
  

net/asterisk-13.x/patches/002-undef-res-ninit.patch

@@ -1,6 +1,6 @@
 --- a/configure.ac
 +++ b/configure.ac
-@@ -1261,7 +1261,6 @@ AC_LINK_IFELSE(
+@@ -1290,7 +1290,6 @@ AC_LINK_IFELSE(
  			#include <resolv.h>],
  			[int foo = res_ninit(NULL);])],
  	AC_MSG_RESULT(yes)

net/asterisk-13.x/patches/004-ifdef-missing-execinfo.patch

@@ -31,7 +31,7 @@
 @@ -114,9 +120,11 @@ struct ast_lock_track {
  	int reentrancy;
  	const char *func[AST_MAX_REENTRANCY];
- 	pthread_t thread[AST_MAX_REENTRANCY];
+ 	pthread_t thread_id[AST_MAX_REENTRANCY];
 +#ifndef __UCLIBC__
  #ifdef HAVE_BKTR
  	struct ast_bt backtrace[AST_MAX_REENTRANCY];

net/asterisk-13.x/patches/040-fix-config-options.patch

@@ -1,12 +0,0 @@
---- a/main/config_options.c
-+++ b/main/config_options.c
-@@ -198,8 +198,8 @@ static int link_option_to_types(struct a
- #ifdef AST_DEVMODE
- 			opt->doc_unavailable = 1;
- #endif
--#endif
- 		}
-+#endif
- 	}
- 	/* The container(s) should hold the only ref to opt */
- 	ao2_ref(opt, -1);

net/asterisk-13.x/patches/051-musl-includes.patch

@@ -1,42 +0,0 @@
---- a/include/asterisk/compat.h
-+++ b/include/asterisk/compat.h
-@@ -68,7 +68,7 @@
- #endif
- 
- #ifndef AST_POLL_COMPAT
--#include <sys/poll.h>
-+#include <poll.h>
- #else
- #include "asterisk/poll-compat.h"
- #endif
---- a/include/asterisk/poll-compat.h
-+++ b/include/asterisk/poll-compat.h
-@@ -83,7 +83,7 @@
- 
- #ifndef AST_POLL_COMPAT
- 
--#include <sys/poll.h>
-+#include <poll.h>
- 
- #define ast_poll(a, b, c) poll(a, b, c)
- 
---- a/main/ast_expr2.c
-+++ b/main/ast_expr2.c
-@@ -93,6 +93,7 @@
- 
- #include "asterisk.h"
- 
-+#include <sys/cdefs.h>
- #include <sys/types.h>
- #include <stdio.h>
- 
---- a/main/ast_expr2.y
-+++ b/main/ast_expr2.y
-@@ -14,6 +14,7 @@
- 
- #include "asterisk.h"
- 
-+#include <sys/cdefs.h>
- #include <sys/types.h>
- #include <stdio.h>
- 

net/asterisk-13.x/patches/052-musl-libcap.patch

@@ -1,7 +1,7 @@
 --- a/configure.ac
 +++ b/configure.ac
 @@ -181,6 +181,9 @@ case "${host_os}" in
-      linux-gnueabi* |  linux-gnuspe)
+      linux-gnu*)
       OSARCH=linux-gnu
       ;;
 +     linux-musl*)
@@ -10,7 +10,7 @@
       kfreebsd*-gnu)
       OSARCH=kfreebsd-gnu
       ;;
-@@ -1373,9 +1376,11 @@ if test "${PBX_BFD}" = "0"; then
+@@ -1414,9 +1417,11 @@ if test "${PBX_BFD}" = "0"; then
    AST_EXT_LIB_CHECK([BFD], [bfd], [bfd_check_format], [bfd.h], [-ldl -liberty -lz])
  fi
  
@@ -26,12 +26,12 @@
  AST_C_DEFINE_CHECK([DAHDI], [DAHDI_DEFAULT_MTU_MRU], [dahdi/user.h], [220])
 --- a/main/Makefile
 +++ b/main/Makefile
-@@ -45,7 +45,7 @@ AST_LIBS+=$(UUID_LIB)
- AST_LIBS+=$(CRYPT_LIB)
- AST_LIBS+=$(AST_CLANG_BLOCKS_LIBS)
+@@ -47,7 +47,7 @@ AST_LIBS+=$(AST_CLANG_BLOCKS_LIBS)
+ AST_LIBS+=$(RT_LIB)
+ AST_LIBS+=$(SYSTEMD_LIB)
  
 -ifneq ($(findstring $(OSARCH), linux-gnu uclinux linux-uclibc kfreebsd-gnu),)
 +ifneq ($(findstring $(OSARCH), linux-gnu uclinux linux-uclibc linux-musl kfreebsd-gnu),)
-   ifneq ($(findstring LOADABLE_MODULES,$(MENUSELECT_CFLAGS)),)
    AST_LIBS+=-ldl
-   endif
+   ifneq (x$(CAP_LIB),x)
+     AST_LIBS+=$(CAP_LIB)