Merge pull request #700 from micmac1/ast16.20

[19.07] bump pjproject, asterisk & chan-lantiq

40 files changed, 886 insertions, 1470 deletions

diff --git a/libs/pjproject/Makefile b/libs/pjproject/Makefile
index c204ed0..fc66027 100644
--- a/libs/pjproject/Makefile
+++ b/libs/pjproject/Makefile
@@ -10,14 +10,15 @@
 include $(TOPDIR)/rules.mk
 
 PKG_NAME:=pjproject
-PKG_VERSION:=2.8
-PKG_RELEASE:=3
-
-PKG_SOURCE:=pjproject-$(PKG_VERSION).tar.bz2
-PKG_SOURCE_URL:=http://www.pjsip.org/release/$(PKG_VERSION)
-PKG_HASH:=503d0bd7f9f13dc1492ac9b71b761b1089851fbb608b9a13996edc3c42006f79
+PKG_VERSION:=2.10
+PKG_RELEASE:=1
+
+# download "vX.Y.tar.gz" as "pjproject-vX.Y.tar.gz"
+PKG_SOURCE_URL_FILE:=$(PKG_VERSION).tar.gz
+PKG_SOURCE:=$(PKG_NAME)-$(PKG_SOURCE_URL_FILE)
+PKG_SOURCE_URL:=https://github.com/pjsip/$(PKG_NAME)/archive
+PKG_HASH:=936a4c5b98601b52325463a397ddf11ab4106c6a7b04f8dc7cdd377efbb597de
 PKG_INSTALL:=1
-PKG_FIXUP:=autoreconf
 
 PKG_LICENSE:=GPL-2.0
 PKG_LICENSE_FILES:=COPYING
@@ -34,7 +35,7 @@ define Package/pjproject/Default
   SECTION:=lib
   CATEGORY:=Libraries
   SUBMENU:=Telephony
-  URL:=http://www.pjsip.org
+  URL:=https://www.pjsip.org
   DEPENDS:=$(CXX_DEPENDS) +libopenssl +libuuid +libpthread
 endef
 
@@ -60,6 +61,7 @@ endef
 CONFIGURE_ARGS+= \
 	$(if $(CONFIG_SOFT_FLOAT),--disable-floating-point) \
 	--disable-bcg729 \
+	--disable-darwin-ssl \
 	--disable-ext-sound \
 	--disable-ffmpeg \
 	--disable-g711-codec \
@@ -92,6 +94,7 @@ CONFIGURE_ARGS+= \
 TARGET_CFLAGS+=$(TARGET_CPPFLAGS)
 
 define Build/Compile
+	$(MAKE) $(PKG_JOBS) -C $(PKG_BUILD_DIR) dep
 	$(MAKE) $(PKG_JOBS) -C $(PKG_BUILD_DIR)
 endef
 
@@ -108,7 +111,8 @@ define Build/InstallDev
 
 	$(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
+	$(INSTALL_DATA) $(PKG_INSTALL_DIR)/usr/lib/pkgconfig/libpjproject.pc \
+		$(1)/usr/lib/pkgconfig
 endef
 
 $(eval $(call PJSIPpackage,libpj,libpj,+librt))
diff --git a/libs/pjproject/patches/001-change_linker.patch b/libs/pjproject/patches/0001-change_linker.patch
index 5fd1503..5fd1503 100644
--- a/libs/pjproject/patches/001-change_linker.patch
+++ b/libs/pjproject/patches/0001-change_linker.patch
diff --git a/libs/pjproject/patches/100-uclibc-linker-unrecognized-options.patch b/libs/pjproject/patches/0002-uclibc-linker-unrecognized-options.patch
index 9930cef..9930cef 100644
--- a/libs/pjproject/patches/100-uclibc-linker-unrecognized-options.patch
+++ b/libs/pjproject/patches/0002-uclibc-linker-unrecognized-options.patch
diff --git a/libs/pjproject/patches/120-non-gnu-pthreads.patch b/libs/pjproject/patches/0003-non-gnu-pthreads.patch
index b44cd70..b44cd70 100644
--- a/libs/pjproject/patches/120-non-gnu-pthreads.patch
+++ b/libs/pjproject/patches/0003-non-gnu-pthreads.patch
diff --git a/libs/pjproject/patches/150-config_site.patch b/libs/pjproject/patches/0004-config_site.patch
index 7e1c897..b5e6d86 100644
--- a/libs/pjproject/patches/150-config_site.patch
+++ b/libs/pjproject/patches/0004-config_site.patch
@@ -58,13 +58,13 @@
 +  Enabling it will result in SEGFAULTS when URIs containing escape sequences are encountered.
 +*/
 +#undef PJSIP_UNESCAPE_IN_PLACE
-+#define PJSIP_MAX_PKT_LEN			32000
++#define PJSIP_MAX_PKT_LEN			65535
 +
 +#undef PJ_TODO
 +#define PJ_TODO(x)
 +
 +/* Defaults too low for WebRTC */
-+#define PJ_ICE_MAX_CAND 32
++#define PJ_ICE_MAX_CAND 64
 +#define PJ_ICE_MAX_CHECKS (PJ_ICE_MAX_CAND * PJ_ICE_MAX_CAND)
 +
 +/* Increase limits to allow more formats */
diff --git a/libs/pjproject/patches/160-remove-hardcoded-lstdc++.patch b/libs/pjproject/patches/0005-remove-hardcoded-lstdc++.patch
index e2bf888..d1db8da 100644
--- a/libs/pjproject/patches/160-remove-hardcoded-lstdc++.patch
+++ b/libs/pjproject/patches/0005-remove-hardcoded-lstdc++.patch
@@ -1,6 +1,6 @@
 --- a/build.mak.in
 +++ b/build.mak.in
-@@ -292,7 +292,6 @@ export APP_LDLIBS := $(PJSUA_LIB_LDLIB)
+@@ -296,7 +296,6 @@ export APP_LDLIBS := $(PJSUA_LIB_LDLIB)
  	$(PJLIB_LDLIB) \
  	@LIBS@
  export APP_LDXXLIBS := $(PJSUA2_LIB_LDLIB) \
diff --git a/libs/pjproject/patches/0006-fix-pkg_config-file.patch b/libs/pjproject/patches/0006-fix-pkg_config-file.patch
new file mode 100644
index 0000000..b8e9391
--- /dev/null
+++ b/libs/pjproject/patches/0006-fix-pkg_config-file.patch
@@ -0,0 +1,25 @@
+--- a/libpjproject.pc.in
++++ b/libpjproject.pc.in
+@@ -2,8 +2,8 @@
+ 
+ prefix=@PREFIX@
+ exec_prefix=${prefix}
+-libdir=@LIBDIR@
+-includedir=@INCLUDEDIR@
++libdir=${exec_prefix}/lib
++includedir=${prefix}/include
+ 
+ Name: libpjproject
+ Description: Multimedia communication library
+--- a/build.mak.in
++++ b/build.mak.in
+@@ -314,8 +314,7 @@ export PJ_LIBXX_FILES := $(APP_LIBXX_FIL
+ # And here are the variables to use if application is using the
+ # library from the install location (i.e. --prefix)
+ export PJ_INSTALL_DIR := @prefix@
+-export PJ_INSTALL_INC_DIR := @includedir@
+ export PJ_INSTALL_LIB_DIR := @libdir@
+-export PJ_INSTALL_CFLAGS := -I$(PJ_INSTALL_INC_DIR) -DPJ_AUTOCONF=1 @CFLAGS@
++export PJ_INSTALL_CFLAGS := -DPJ_AUTOCONF=1 @CFLAGS@
+ export PJ_INSTALL_CXXFLAGS := @CXXFLAGS@ $(PJ_INSTALL_CFLAGS)
+ export PJ_INSTALL_LDFLAGS := -L$(PJ_INSTALL_LIB_DIR) $(APP_LDLIBS)
diff --git a/libs/pjproject/patches/0011-sip_inv_patch.patch b/libs/pjproject/patches/0011-sip_inv_patch.patch
new file mode 100644
index 0000000..c410b9a
--- /dev/null
+++ b/libs/pjproject/patches/0011-sip_inv_patch.patch
@@ -0,0 +1,37 @@
+commit c3c1bf45cae2a35003aa16c267d59f97027f9c5e
+Author: Kevin Harwell <kharwell@digium.com>
+Date:   Thu Jun 11 11:11:13 2020 -0500
+
+    sip_inv - fix invite session ref count crash
+    
+    Ensure the session's ref count is only decremented under proper conditons.
+    
+    For more details see the following issue report:
+    https://github.com/pjsip/pjproject/issues/2443
+    
+    Patch supplied by sauwming
+
+--- a/pjsip/src/pjsip-ua/sip_inv.c
++++ b/pjsip/src/pjsip-ua/sip_inv.c
+@@ -323,9 +323,19 @@ static void inv_set_state(pjsip_inv_sess
+ 	(*mod_inv.cb.on_state_changed)(inv, e);
+     pjsip_inv_dec_ref(inv);
+ 
+-    /* Only decrement when previous state is not already DISCONNECTED */
++    /* The above callback may change the state, so we need to be careful here
++     * and only decrement inv under the following conditions:
++     * 1. If the state parameter is DISCONNECTED, and previous state is not
++     *    already DISCONNECTED.
++     *    This is to make sure that dec_ref() is not called more than once.
++     * 2. If current state is PJSIP_INV_STATE_DISCONNECTED.
++     *    This is to make sure that dec_ref() is not called if user restarts
++     *    inv within the callback. Note that this check must be last since
++     *    inv may have already been destroyed.
++     */
+     if (state == PJSIP_INV_STATE_DISCONNECTED &&
+-	prev_state != PJSIP_INV_STATE_DISCONNECTED) 
++	prev_state != PJSIP_INV_STATE_DISCONNECTED &&
++	inv->state == PJSIP_INV_STATE_DISCONNECTED) 
+     {
+ 	pjsip_inv_dec_ref(inv);
+     }
diff --git a/libs/pjproject/patches/0020-pjlib_cancel_timer_0.patch b/libs/pjproject/patches/0020-pjlib_cancel_timer_0.patch
new file mode 100644
index 0000000..3fe6294
--- /dev/null
+++ b/libs/pjproject/patches/0020-pjlib_cancel_timer_0.patch
@@ -0,0 +1,35 @@
+commit 40dd48d10911f4ff9b8dfbf16428fbc9acc434ba
+Author: Riza Sulistyo <trengginas@users.noreply.github.com>
+Date:   Thu Jul 9 17:47:24 2020 +0700
+
+    Modify timer_id check on cancel() (#2463)
+    
+    * modify timer_id check on cancel().
+    
+    * modification based on comments.
+
+--- a/pjlib/include/pj/timer.h
++++ b/pjlib/include/pj/timer.h
+@@ -120,7 +120,10 @@ typedef struct pj_timer_entry
+ 
+     /** 
+      * Internal unique timer ID, which is assigned by the timer heap. 
+-     * Application should not touch this ID.
++     * Positive values indicate that the timer entry is running, 
++     * while -1 means that it's not. Any other value may indicate that it 
++     * hasn't been properly initialised or is in a bad state.
++     * Application should not touch this ID. 
+      */
+     pj_timer_id_t _timer_id;
+ 
+--- a/pjlib/src/pj/timer.c
++++ b/pjlib/src/pj/timer.c
+@@ -535,7 +535,7 @@ static int cancel( pj_timer_heap_t *ht,
+     PJ_CHECK_STACK();
+ 
+     // Check to see if the timer_id is out of range
+-    if (entry->_timer_id < 0 || (pj_size_t)entry->_timer_id > ht->max_size) {
++    if (entry->_timer_id < 1 || (pj_size_t)entry->_timer_id >= ht->max_size) {
+ 	entry->_timer_id = -1;
+     	return 0;
+     }
diff --git a/libs/pjproject/patches/0050-fix-race-parallel-build.patch b/libs/pjproject/patches/0050-fix-race-parallel-build.patch
new file mode 100644
index 0000000..3a48d41
--- /dev/null
+++ b/libs/pjproject/patches/0050-fix-race-parallel-build.patch
@@ -0,0 +1,68 @@
+From 78683646c8bc670ec730a42494e075f671a08e28 Mon Sep 17 00:00:00 2001
+From: Guido Falsi <mad@madpilot.net>
+Date: Mon, 11 May 2020 08:50:39 +0200
+Subject: [PATCH] Fix race condition in parallel builds (#2426)
+
+* Some targets residing in `OBJDIRS` are missing a dependency on that directory, which results in a race condition, causing build to fail sometimes due to the directory not existing when running parallel builds.
+
+* The `PJSUA_LIB` variable is not defined anywhere, resulting in an empty value, and no correct dependency on the pjsua shared library for `pjsua2`. The correct variable seems to be `PJSUA_LIB_LIB`, defined at the start of this same `Makefile`.
+---
+ build/rules.mak      | 12 ++++++------
+ pjsip/build/Makefile |  2 +-
+ 2 files changed, 7 insertions(+), 7 deletions(-)
+
+--- a/build/rules.mak
++++ b/build/rules.mak
+@@ -129,7 +129,7 @@ endif
+ $(OBJDIR)/$(app).o: $(OBJDIRS) $(OBJS)
+ 	$(CROSS_COMPILE)ld -r -o $@ $(OBJS)
+ 
+-$(OBJDIR)/$(app).ko: $(OBJDIR)/$(app).o
++$(OBJDIR)/$(app).ko: $(OBJDIR)/$(app).o | $(OBJDIRS)
+ 	@echo Creating kbuild Makefile...
+ 	@echo "# Our module name:" > $(OBJDIR)/Makefile
+ 	@echo 'obj-m += $(app).o' >> $(OBJDIR)/Makefile
+@@ -154,27 +154,27 @@ $(OBJDIR)/$(app).ko: $(OBJDIR)/$(app).o
+ ../lib/$(app).ko: $(LIB) $(OBJDIR)/$(app).ko
+ 	cp $(OBJDIR)/$(app).ko ../lib
+ 
+-$(OBJDIR)/%$(OBJEXT): $(SRCDIR)/%.m
++$(OBJDIR)/%$(OBJEXT): $(SRCDIR)/%.m | $(OBJDIRS)
+ 	$(CC) $($(APP)_CFLAGS) \
+ 		$(CC_OUT)$(subst /,$(HOST_PSEP),$@) \
+ 		$(subst /,$(HOST_PSEP),$<) 
+ 
+-$(OBJDIR)/%$(OBJEXT): $(SRCDIR)/%.c
++$(OBJDIR)/%$(OBJEXT): $(SRCDIR)/%.c | $(OBJDIRS)
+ 	$(CC) $($(APP)_CFLAGS) \
+ 		$(CC_OUT)$(subst /,$(HOST_PSEP),$@) \
+ 		$(subst /,$(HOST_PSEP),$<) 
+ 
+-$(OBJDIR)/%$(OBJEXT): $(SRCDIR)/%.S
++$(OBJDIR)/%$(OBJEXT): $(SRCDIR)/%.S | $(OBJDIRS)
+ 	$(CC) $($(APP)_CFLAGS) \
+ 		$(CC_OUT)$(subst /,$(HOST_PSEP),$@) \
+ 		$(subst /,$(HOST_PSEP),$<) 
+ 
+-$(OBJDIR)/%$(OBJEXT): $(SRCDIR)/%.cpp
++$(OBJDIR)/%$(OBJEXT): $(SRCDIR)/%.cpp | $(OBJDIRS)
+ 	$(CXX) $($(APP)_CXXFLAGS) \
+ 		$(CC_OUT)$(subst /,$(HOST_PSEP),$@) \
+ 		$(subst /,$(HOST_PSEP),$<)
+ 
+-$(OBJDIR)/%$(OBJEXT): $(SRCDIR)/%.cc
++$(OBJDIR)/%$(OBJEXT): $(SRCDIR)/%.cc | $(OBJDIRS)
+ 	$(CXX) $($(APP)_CXXFLAGS) \
+ 		$(CC_OUT)$(subst /,$(HOST_PSEP),$@) \
+ 		$(subst /,$(HOST_PSEP),$<)
+--- a/pjsip/build/Makefile
++++ b/pjsip/build/Makefile
+@@ -261,7 +261,7 @@ $(PJSUA_LIB_LIB) $(PJSUA_LIB_SONAME): $(
+ 
+ pjsua2-lib: $(PJSUA2_LIB_LIB)
+ $(PJSUA2_LIB_SONAME): $(PJSUA2_LIB_LIB)
+-$(PJSUA2_LIB_LIB) $(PJSUA2_LIB_SONAME): $(PJSUA_LIB) $(PSJUA_LIB_SONAME) $(PJSIP_LIB) $(PJSIP_SONAME) $(PJSIP_SIMPLE_LIB) $(PJSIP_SIMPLE_SONAME) $(PJSIP_UA_LIB) $(PJSIP_UA_SONAME)
++$(PJSUA2_LIB_LIB) $(PJSUA2_LIB_SONAME): $(PJSUA_LIB_LIB) $(PJSUA_LIB_SONAME) $(PJSIP_LIB) $(PJSIP_SONAME) $(PJSIP_SIMPLE_LIB) $(PJSIP_SIMPLE_SONAME) $(PJSIP_UA_LIB) $(PJSIP_UA_SONAME)
+ 	$(MAKE) -f $(RULES_MAK) APP=PJSUA2_LIB app=pjsua2-lib $(subst /,$(HOST_PSEP),$(LIBDIR)/$@)
+ 
+ pjsip-test: $(TEST_EXE)
diff --git a/libs/pjproject/patches/0060-clone-sdp-for-sip-timer-refresh-invite.patch b/libs/pjproject/patches/0060-clone-sdp-for-sip-timer-refresh-invite.patch
new file mode 100644
index 0000000..0fda06d
--- /dev/null
+++ b/libs/pjproject/patches/0060-clone-sdp-for-sip-timer-refresh-invite.patch
@@ -0,0 +1,27 @@
+--- a/pjmedia/src/pjmedia/sdp_neg.c
++++ b/pjmedia/src/pjmedia/sdp_neg.c
+@@ -906,7 +906,7 @@ static pj_status_t process_m_answer( pj_
+  * after receiving remote answer.
+  */
+ static pj_status_t process_answer(pj_pool_t *pool,
+-				  pjmedia_sdp_session *offer,
++				  pjmedia_sdp_session *local_offer,
+ 				  pjmedia_sdp_session *answer,
+ 				  pj_bool_t allow_asym,
+ 				  pjmedia_sdp_session **p_active)
+@@ -914,10 +914,14 @@ static pj_status_t process_answer(pj_poo
+     unsigned omi = 0; /* Offer media index */
+     unsigned ami = 0; /* Answer media index */
+     pj_bool_t has_active = PJ_FALSE;
++    pjmedia_sdp_session *offer;
+     pj_status_t status;
+ 
+     /* Check arguments. */
+-    PJ_ASSERT_RETURN(pool && offer && answer && p_active, PJ_EINVAL);
++    PJ_ASSERT_RETURN(pool && local_offer && answer && p_active, PJ_EINVAL);
++
++    /* Duplicate local offer SDP. */
++    offer = pjmedia_sdp_session_clone(pool, local_offer);
+ 
+     /* Check that media count match between offer and answer */
+     // Ticket #527, different media count is allowed for more interoperability,
diff --git a/libs/pjproject/patches/0070-fix-incorrect-copying-when-creating-cancel.patch b/libs/pjproject/patches/0070-fix-incorrect-copying-when-creating-cancel.patch
new file mode 100644
index 0000000..c8e4b4c
--- /dev/null
+++ b/libs/pjproject/patches/0070-fix-incorrect-copying-when-creating-cancel.patch
@@ -0,0 +1,32 @@
+From ce18018cc17bef8f80c08686e3a7b28384ef3ba5 Mon Sep 17 00:00:00 2001
+From: sauwming <ming@teluu.com>
+Date: Mon, 12 Oct 2020 13:31:25 +0800
+Subject: [PATCH] Fix incorrect copying of destination info when creating
+ CANCEL (#2546)
+
+---
+ pjsip/src/pjsip/sip_util.c | 10 +++++-----
+ 1 file changed, 5 insertions(+), 5 deletions(-)
+
+--- a/pjsip/src/pjsip/sip_util.c
++++ b/pjsip/src/pjsip/sip_util.c
+@@ -779,14 +779,14 @@ PJ_DEF(pj_status_t) pjsip_endpt_create_c
+ 	    pjsip_hdr_clone(cancel_tdata->pool, req_tdata->saved_strict_route);
+     }
+ 
+-    /* Copy the destination host name from the original request */
+-    pj_strdup(cancel_tdata->pool, &cancel_tdata->dest_info.name,
+-	      &req_tdata->dest_info.name);
+-
+-    /* Finally copy the destination info from the original request */
++    /* Copy the destination info from the original request */
+     pj_memcpy(&cancel_tdata->dest_info, &req_tdata->dest_info,
+ 	      sizeof(req_tdata->dest_info));
+ 
++    /* Finally, copy the destination host name from the original request */
++    pj_strdup(cancel_tdata->pool, &cancel_tdata->dest_info.name,
++	      &req_tdata->dest_info.name);
++
+     /* Done.
+      * Return the transmit buffer containing the CANCEL request.
+      */
diff --git a/libs/pjproject/patches/240-fix-sdp-neg-modify-local-offer.patch b/libs/pjproject/patches/0080-fix-sdp-neg-modify-local-offer.patch
index c27a489..889d012 100644
--- a/libs/pjproject/patches/240-fix-sdp-neg-modify-local-offer.patch
+++ b/libs/pjproject/patches/0080-fix-sdp-neg-modify-local-offer.patch
@@ -1,8 +1,6 @@
-diff --git a/pjmedia/src/pjmedia/sdp_neg.c b/pjmedia/src/pjmedia/sdp_neg.c
-index 3b85b4273..a14009662 100644
 --- a/pjmedia/src/pjmedia/sdp_neg.c
 +++ b/pjmedia/src/pjmedia/sdp_neg.c
-@@ -304,7 +304,6 @@ PJ_DEF(pj_status_t) pjmedia_sdp_neg_modify_local_offer2(
+@@ -304,7 +304,6 @@ PJ_DEF(pj_status_t) pjmedia_sdp_neg_modi
  {
      pjmedia_sdp_session *new_offer;
      pjmedia_sdp_session *old_offer;
@@ -10,7 +8,7 @@ index 3b85b4273..a14009662 100644
      unsigned oi; /* old offer media index */
      pj_status_t status;
  
-@@ -323,8 +322,19 @@ PJ_DEF(pj_status_t) pjmedia_sdp_neg_modify_local_offer2(
+@@ -323,8 +322,19 @@ PJ_DEF(pj_status_t) pjmedia_sdp_neg_modi
      /* Change state to STATE_LOCAL_OFFER */
      neg->state = PJMEDIA_SDP_NEG_STATE_LOCAL_OFFER;
  
diff --git a/libs/pjproject/patches/0090-Skip-unsupported-digest-algorithm-2408.patch b/libs/pjproject/patches/0090-Skip-unsupported-digest-algorithm-2408.patch
new file mode 100644
index 0000000..c779f67
--- /dev/null
+++ b/libs/pjproject/patches/0090-Skip-unsupported-digest-algorithm-2408.patch
@@ -0,0 +1,201 @@
+From bdbeb7c4b2b11efc2e59f5dee7aa4360a2bc9fff Mon Sep 17 00:00:00 2001
+From: sauwming <ming@teluu.com>
+Date: Thu, 22 Apr 2021 14:03:28 +0800
+Subject: [PATCH 90/90] Skip unsupported digest algorithm (#2408)
+
+Co-authored-by: Nanang Izzuddin <nanang@teluu.com>
+---
+ pjsip/src/pjsip/sip_auth_client.c             | 32 +++++--
+ tests/pjsua/scripts-sipp/uas-auth-two-algo.py |  7 ++
+ .../pjsua/scripts-sipp/uas-auth-two-algo.xml  | 83 +++++++++++++++++++
+ 3 files changed, 117 insertions(+), 5 deletions(-)
+ create mode 100644 tests/pjsua/scripts-sipp/uas-auth-two-algo.py
+ create mode 100644 tests/pjsua/scripts-sipp/uas-auth-two-algo.xml
+
+--- a/pjsip/src/pjsip/sip_auth_client.c
++++ b/pjsip/src/pjsip/sip_auth_client.c
+@@ -1042,7 +1042,7 @@ static pj_status_t process_auth( pj_pool
+     pjsip_hdr *hdr;
+     pj_status_t status;
+ 
+-    /* See if we have sent authorization header for this realm */
++    /* See if we have sent authorization header for this realm (and scheme) */
+     hdr = tdata->msg->hdr.next;
+     while (hdr != &tdata->msg->hdr) {
+ 	if ((hchal->type == PJSIP_H_WWW_AUTHENTICATE &&
+@@ -1052,7 +1052,8 @@ static pj_status_t process_auth( pj_pool
+ 	{
+ 	    sent_auth = (pjsip_authorization_hdr*) hdr;
+ 	    if (pj_stricmp(&hchal->challenge.common.realm,
+-			   &sent_auth->credential.common.realm )==0)
++			   &sent_auth->credential.common.realm)==0 &&
++		pj_stricmp(&hchal->scheme, &sent_auth->scheme)==0)
+ 	    {
+ 		/* If this authorization has empty response, remove it. */
+ 		if (pj_stricmp(&sent_auth->scheme, &pjsip_DIGEST_STR)==0 &&
+@@ -1062,6 +1063,14 @@ static pj_status_t process_auth( pj_pool
+ 		    hdr = hdr->next;
+ 		    pj_list_erase(sent_auth);
+ 		    continue;
++		} else
++		if (pj_stricmp(&sent_auth->scheme, &pjsip_DIGEST_STR)==0 &&
++		    pj_stricmp(&sent_auth->credential.digest.algorithm,
++		               &hchal->challenge.digest.algorithm)!=0)
++		{
++		    /* Same 'digest' scheme but different algo */
++		    hdr = hdr->next;
++		    continue;
+ 		} else {
+ 		    /* Found previous authorization attempt */
+ 		    break;
+@@ -1155,9 +1164,10 @@ PJ_DEF(pj_status_t) pjsip_auth_clt_reini
+ {
+     pjsip_tx_data *tdata;
+     const pjsip_hdr *hdr;
+-    unsigned chal_cnt;
++    unsigned chal_cnt, auth_cnt;
+     pjsip_via_hdr *via;
+     pj_status_t status;
++    pj_status_t last_auth_err;
+ 
+     PJ_ASSERT_RETURN(sess && rdata && old_request && new_request,
+ 		     PJ_EINVAL);
+@@ -1178,6 +1188,8 @@ PJ_DEF(pj_status_t) pjsip_auth_clt_reini
+      */
+     hdr = rdata->msg_info.msg->hdr.next;
+     chal_cnt = 0;
++    auth_cnt = 0;
++    last_auth_err = PJSIP_EAUTHNOAUTH;
+     while (hdr != &rdata->msg_info.msg->hdr) {
+ 	pjsip_cached_auth *cached_auth;
+ 	const pjsip_www_authenticate_hdr *hchal;
+@@ -1222,8 +1234,13 @@ PJ_DEF(pj_status_t) pjsip_auth_clt_reini
+ 	 */
+ 	status = process_auth(tdata->pool, hchal, tdata->msg->line.req.uri,
+ 			      tdata, sess, cached_auth, &hauth);
+-	if (status != PJ_SUCCESS)
+-	    return status;
++	if (status != PJ_SUCCESS) {
++	    last_auth_err = status;
++
++	    /* Process next header. */
++	    hdr = hdr->next;
++	    continue;
++	}
+ 
+ 	if (pj_pool_get_used_size(cached_auth->pool) >
+ 	    PJSIP_AUTH_CACHED_POOL_MAX_SIZE) 
+@@ -1236,12 +1253,17 @@ PJ_DEF(pj_status_t) pjsip_auth_clt_reini
+ 
+ 	/* Process next header. */
+ 	hdr = hdr->next;
++	auth_cnt++;
+     }
+ 
+     /* Check if challenge is present */
+     if (chal_cnt == 0)
+ 	return PJSIP_EAUTHNOCHAL;
+ 
++    /* Check if any authorization header has been created */
++    if (auth_cnt == 0)
++	return last_auth_err;
++
+     /* Remove branch param in Via header. */
+     via = (pjsip_via_hdr*) pjsip_msg_find_hdr(tdata->msg, PJSIP_H_VIA, NULL);
+     via->branch_param.slen = 0;
+--- /dev/null
++++ b/tests/pjsua/scripts-sipp/uas-auth-two-algo.py
+@@ -0,0 +1,7 @@
++# $Id$
++#
++import inc_const as const
++
++PJSUA = ["--null-audio --max-calls=1 --id=sip:a@localhost --username=a --realm=* --registrar=$SIPP_URI"]
++
++PJSUA_EXPECTS = [[0, "registration success", ""]]
+--- /dev/null
++++ b/tests/pjsua/scripts-sipp/uas-auth-two-algo.xml
+@@ -0,0 +1,83 @@
++<?xml version="1.0" encoding="ISO-8859-1" ?>
++<!DOCTYPE scenario SYSTEM "sipp.dtd">
++
++<scenario name="Basic UAS responder">
++  <recv request="REGISTER" crlf="true">
++  </recv>
++
++  <send>
++    <![CDATA[
++      SIP/2.0 100 Trying
++      [last_Via:];received=1.1.1.1;rport=1111
++      [last_From:]
++      [last_To:];tag=[call_number]
++      [last_Call-ID:]
++      [last_CSeq:]
++      Content-Length: 0
++    ]]>
++  </send>
++
++  <send>
++    <![CDATA[
++      SIP/2.0 401 Unauthorized
++      [last_Via:];received=1.1.1.1;rport=1111
++      [last_From:]
++      [last_To:];tag=[call_number]
++      [last_Call-ID:]
++      [last_CSeq:]
++      WWW-Authenticate: Digest realm="sip.linphone.org", nonce="PARV4gAAAADgw3asAADW8zsi5BEAAAAA", opaque="+GNywA==", algorithm=SHA-256, qop="auth"
++      WWW-Authenticate: Digest realm="sip.linphone.org", nonce="PARV4gAAAADgw3asAADW8zsi5BEAAAAA", opaque="+GNywA==", algorithm=MD5, qop="auth"
++      WWW-Authenticate: Digest realm="sip.linphone.org", nonce="PARV4gAAAADgw3asAADW8zsi5BEAAAAA", opaque="+GNywA==", algorithm=MD2, qop="auth"
++      Content-Length: 0
++    ]]>
++  </send>
++
++  <recv request="REGISTER" crlf="true">
++    <action>
++      <ereg regexp=".*"
++            search_in="hdr"
++	    header="Authorization:"
++	    assign_to="have_auth" />
++    </action>
++  </recv>
++
++  <nop next="resp_okay" test="have_auth" />
++  
++  <send next="end">
++    <![CDATA[
++      SIP/2.0 403 no auth
++      [last_Via:];received=1.1.1.1;rport=1111
++      [last_From:]
++      [last_To:];tag=[call_number]
++      [last_Call-ID:]
++      [last_CSeq:]
++      [last_Contact:]
++      Content-Length: 0
++    ]]>
++  </send>
++
++  <label id="resp_okay" />
++  
++  <send>
++    <![CDATA[
++      SIP/2.0 200 OK
++      [last_Via:];received=1.1.1.1;rport=1111
++      [last_From:]
++      [last_To:];tag=[call_number]
++      [last_Call-ID:]
++      [last_CSeq:]
++      [last_Contact:]
++      Content-Length: 0
++    ]]>
++  </send>
++
++  <label id="end" />
++
++  <!-- definition of the response time repartition table (unit is ms)   -->
++  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
++
++  <!-- definition of the call length repartition table (unit is ms)     -->
++  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
++
++</scenario>
++
diff --git a/libs/pjproject/patches/0100-fix-double-stun-free.patch b/libs/pjproject/patches/0100-fix-double-stun-free.patch
new file mode 100644
index 0000000..f1691b9
--- /dev/null
+++ b/libs/pjproject/patches/0100-fix-double-stun-free.patch
@@ -0,0 +1,78 @@
+commit f0ff5817d0647bdecd1ec99488db9378e304cf83
+Author: sauwming <ming@teluu.com>
+Date:   Mon May 17 09:56:27 2021 +0800
+
+    Fix double free of stun session (#2709)
+
+--- a/pjnath/include/pjnath/stun_session.h
++++ b/pjnath/include/pjnath/stun_session.h
+@@ -341,6 +341,7 @@ struct pj_stun_tx_data
+     pj_pool_t		*pool;		/**< Pool.			    */
+     pj_stun_session	*sess;		/**< The STUN session.		    */
+     pj_stun_msg		*msg;		/**< The STUN message.		    */
++    pj_bool_t		 is_destroying; /**< Is destroying?		    */
+ 
+     void		*token;		/**< The token.			    */
+ 
+--- a/pjnath/src/pjnath/stun_session.c
++++ b/pjnath/src/pjnath/stun_session.c
+@@ -167,16 +167,27 @@ static void tdata_on_destroy(void *arg)
+ {
+     pj_stun_tx_data *tdata = (pj_stun_tx_data*)arg;
+ 
++    if (tdata->grp_lock) {
++	pj_grp_lock_dec_ref(tdata->sess->grp_lock);
++    }
++
+     pj_pool_safe_release(&tdata->pool);
+ }
+ 
+ static void destroy_tdata(pj_stun_tx_data *tdata, pj_bool_t force)
+ {
+-    TRACE_((THIS_FILE, "tdata %p destroy request, force=%d, tsx=%p", tdata,
+-	    force, tdata->client_tsx));
++    TRACE_((THIS_FILE,
++	    "tdata %p destroy request, force=%d, tsx=%p, destroying=%d",
++	    tdata, force, tdata->client_tsx, tdata->is_destroying));
++
++    /* Just return if destroy has been requested before */
++    if (tdata->is_destroying)
++	return;
+ 
+     /* STUN session may have been destroyed, except when tdata is cached. */
+ 
++    tdata->is_destroying = PJ_TRUE;
++
+     if (tdata->res_timer.id != PJ_FALSE) {
+ 	pj_timer_heap_cancel_if_active(tdata->sess->cfg->timer_heap,
+ 				       &tdata->res_timer, PJ_FALSE);
+@@ -189,7 +200,6 @@ static void destroy_tdata(pj_stun_tx_dat
+ 	    pj_stun_client_tsx_set_data(tdata->client_tsx, NULL);
+ 	}
+ 	if (tdata->grp_lock) {
+-	    pj_grp_lock_dec_ref(tdata->sess->grp_lock);
+ 	    pj_grp_lock_dec_ref(tdata->grp_lock);
+ 	} else {
+ 	    tdata_on_destroy(tdata);
+@@ -200,11 +210,11 @@ static void destroy_tdata(pj_stun_tx_dat
+ 	    /* "Probably" this is to absorb retransmission */
+ 	    pj_time_val delay = {0, 300};
+ 	    pj_stun_client_tsx_schedule_destroy(tdata->client_tsx, &delay);
++	    tdata->is_destroying = PJ_FALSE;
+ 
+ 	} else {
+ 	    pj_list_erase(tdata);
+ 	    if (tdata->grp_lock) {
+-		pj_grp_lock_dec_ref(tdata->sess->grp_lock);
+ 		pj_grp_lock_dec_ref(tdata->grp_lock);
+ 	    } else {
+ 		tdata_on_destroy(tdata);
+@@ -238,7 +248,7 @@ static void on_cache_timeout(pj_timer_he
+     sess = tdata->sess;
+ 
+     pj_grp_lock_acquire(sess->grp_lock);
+-    if (sess->is_destroying) {
++    if (sess->is_destroying || tdata->is_destroying) {
+ 	pj_grp_lock_release(sess->grp_lock);
+ 	return;
+     }
diff --git a/libs/pjproject/patches/0110-tls-parent-listener-destroyed.patch b/libs/pjproject/patches/0110-tls-parent-listener-destroyed.patch
new file mode 100644
index 0000000..e496fe4
--- /dev/null
+++ b/libs/pjproject/patches/0110-tls-parent-listener-destroyed.patch
@@ -0,0 +1,162 @@
+From bb92c97ea512aa0ef316c9b2335c7d57b84dfc9a Mon Sep 17 00:00:00 2001
+From: Nanang Izzuddin <nanang@teluu.com>
+Date: Wed, 16 Jun 2021 12:12:35 +0700
+Subject: [PATCH 1/2] - Avoid SSL socket parent/listener getting destroyed
+ during handshake by increasing parent's reference count. - Add missing SSL
+ socket close when the newly accepted SSL socket is discarded in SIP TLS
+ transport.
+
+---
+ pjlib/src/pj/ssl_sock_imp_common.c  | 44 +++++++++++++++++++++--------
+ pjsip/src/pjsip/sip_transport_tls.c | 23 ++++++++++++++-
+ 2 files changed, 55 insertions(+), 12 deletions(-)
+
+--- a/pjlib/src/pj/ssl_sock_imp_common.c
++++ b/pjlib/src/pj/ssl_sock_imp_common.c
+@@ -224,6 +224,8 @@ static pj_bool_t on_handshake_complete(p
+ 
+     /* Accepting */
+     if (ssock->is_server) {
++	pj_bool_t ret = PJ_TRUE;
++
+ 	if (status != PJ_SUCCESS) {
+ 	    /* Handshake failed in accepting, destroy our self silently. */
+ 
+@@ -241,6 +243,12 @@ static pj_bool_t on_handshake_complete(p
+ 		      status);
+ 	    }
+ 
++	    /* Decrement ref count of parent */
++	    if (ssock->parent->param.grp_lock) {
++		pj_grp_lock_dec_ref(ssock->parent->param.grp_lock);
++		ssock->parent = NULL;
++	    }
++
+ 	    /* Originally, this is a workaround for ticket #985. However,
+ 	     * a race condition may occur in multiple worker threads
+ 	     * environment when we are destroying SSL objects while other
+@@ -284,23 +292,29 @@ static pj_bool_t on_handshake_complete(p
+ 
+ 	    return PJ_FALSE;
+ 	}
++
+ 	/* Notify application the newly accepted SSL socket */
+ 	if (ssock->param.cb.on_accept_complete2) {
+-	    pj_bool_t ret;
+ 	    ret = (*ssock->param.cb.on_accept_complete2) 
+ 		    (ssock->parent, ssock, (pj_sockaddr_t*)&ssock->rem_addr, 
+ 		    pj_sockaddr_get_len((pj_sockaddr_t*)&ssock->rem_addr), 
+ 		    status);
+-	    if (ret == PJ_FALSE)
+-		return PJ_FALSE;	
+ 	} else if (ssock->param.cb.on_accept_complete) {
+-	    pj_bool_t ret;
+ 	    ret = (*ssock->param.cb.on_accept_complete)
+ 		      (ssock->parent, ssock, (pj_sockaddr_t*)&ssock->rem_addr,
+ 		       pj_sockaddr_get_len((pj_sockaddr_t*)&ssock->rem_addr));
+-	    if (ret == PJ_FALSE)
+-		return PJ_FALSE;
+ 	}
++
++	/* Decrement ref count of parent and reset parent (we don't need it
++	 * anymore, right?).
++	 */
++	if (ssock->parent->param.grp_lock) {
++	    pj_grp_lock_dec_ref(ssock->parent->param.grp_lock);
++	    ssock->parent = NULL;
++	}
++
++	if (ret == PJ_FALSE)
++	    return PJ_FALSE;
+     }
+ 
+     /* Connecting */
+@@ -864,9 +878,13 @@ static pj_bool_t asock_on_accept_complet
+     if (status != PJ_SUCCESS)
+ 	goto on_return;
+ 
++    /* Set parent and add ref count (avoid parent destroy during handshake) */
++    ssock->parent = ssock_parent;
++    if (ssock->parent->param.grp_lock)
++	pj_grp_lock_add_ref(ssock->parent->param.grp_lock);
++
+     /* Update new SSL socket attributes */
+     ssock->sock = newsock;
+-    ssock->parent = ssock_parent;
+     ssock->is_server = PJ_TRUE;
+     if (ssock_parent->cert) {
+ 	status = pj_ssl_sock_set_certificate(ssock, ssock->pool, 
+@@ -913,16 +931,20 @@ static pj_bool_t asock_on_accept_complet
+     ssock->asock_rbuf = (void**)pj_pool_calloc(ssock->pool, 
+ 					       ssock->param.async_cnt,
+ 					       sizeof(void*));
+-    if (!ssock->asock_rbuf)
+-        return PJ_ENOMEM;
++    if (!ssock->asock_rbuf) {
++		status = PJ_ENOMEM;
++		goto on_return;
++	}
+ 
+     for (i = 0; i<ssock->param.async_cnt; ++i) {
+-	ssock->asock_rbuf[i] = (void*) pj_pool_alloc(
++		ssock->asock_rbuf[i] = (void*) pj_pool_alloc(
+ 					    ssock->pool, 
+ 					    ssock->param.read_buffer_size + 
+ 					    sizeof(read_data_t*));
+-        if (!ssock->asock_rbuf[i])
+-            return PJ_ENOMEM;
++		if (!ssock->asock_rbuf[i]) {
++			status = PJ_ENOMEM;
++			goto on_return;
++		}
+     }
+ 
+     /* Create active socket */
+--- a/pjsip/src/pjsip/sip_transport_tls.c
++++ b/pjsip/src/pjsip/sip_transport_tls.c
+@@ -1325,9 +1325,26 @@ static pj_bool_t on_accept_complete2(pj_
+     PJ_UNUSED_ARG(src_addr_len);
+ 
+     listener = (struct tls_listener*) pj_ssl_sock_get_user_data(ssock);
++    if (!listener) {
++	/* Listener already destroyed, e.g: after TCP accept but before SSL
++	 * handshake is completed.
++	 */
++	if (new_ssock && accept_status == PJ_SUCCESS) {
++	    /* Close the SSL socket if the accept op is successful */
++	    PJ_LOG(4,(THIS_FILE,
++		      "Incoming TLS connection from %s (sock=%d) is discarded "
++		      "because listener is already destroyed",
++		      pj_sockaddr_print(src_addr, addr, sizeof(addr), 3),
++		      new_ssock));
++
++	    pj_ssl_sock_close(new_ssock);
++	}
++
++	return PJ_FALSE;
++    }
+ 
+     if (accept_status != PJ_SUCCESS) {
+-	if (listener && listener->tls_setting.on_accept_fail_cb) {
++	if (listener->tls_setting.on_accept_fail_cb) {
+ 	    pjsip_tls_on_accept_fail_param param;
+ 	    pj_ssl_sock_info ssi;
+ 
+@@ -1350,6 +1367,8 @@ static pj_bool_t on_accept_complete2(pj_
+     PJ_ASSERT_RETURN(new_ssock, PJ_TRUE);
+ 
+     if (!listener->is_registered) {
++	pj_ssl_sock_close(new_ssock);
++
+ 	if (listener->tls_setting.on_accept_fail_cb) {
+ 	    pjsip_tls_on_accept_fail_param param;
+ 	    pj_bzero(&param, sizeof(param));
+@@ -1401,6 +1420,8 @@ static pj_bool_t on_accept_complete2(pj_
+ 			 ssl_info.grp_lock, &tls);
+     
+     if (status != PJ_SUCCESS) {
++	pj_ssl_sock_close(new_ssock);
++
+ 	if (listener->tls_setting.on_accept_fail_cb) {
+ 	    pjsip_tls_on_accept_fail_param param;
+ 	    pj_bzero(&param, sizeof(param));
diff --git a/libs/pjproject/patches/0111-ssl-premature-destroy.patch b/libs/pjproject/patches/0111-ssl-premature-destroy.patch
new file mode 100644
index 0000000..3345393
--- /dev/null
+++ b/libs/pjproject/patches/0111-ssl-premature-destroy.patch
@@ -0,0 +1,132 @@
+From 68c69f516f95df1faa42e5647e9ce7cfdc41ac38 Mon Sep 17 00:00:00 2001
+From: Nanang Izzuddin <nanang@teluu.com>
+Date: Wed, 16 Jun 2021 12:15:29 +0700
+Subject: [PATCH 2/2] - Fix silly mistake: accepted active socket created
+ without group lock in SSL socket. - Replace assertion with normal validation
+ check of SSL socket instance in OpenSSL verification callback (verify_cb())
+ to avoid crash, e.g: if somehow race condition with SSL socket destroy
+ happens or OpenSSL application data index somehow gets corrupted.
+
+---
+ pjlib/src/pj/ssl_sock_imp_common.c |  3 +-
+ pjlib/src/pj/ssl_sock_ossl.c       | 45 +++++++++++++++++++++++++-----
+ 2 files changed, 40 insertions(+), 8 deletions(-)
+
+--- a/pjlib/src/pj/ssl_sock_imp_common.c
++++ b/pjlib/src/pj/ssl_sock_imp_common.c
+@@ -949,6 +949,7 @@ static pj_bool_t asock_on_accept_complet
+ 
+     /* Create active socket */
+     pj_activesock_cfg_default(&asock_cfg);
++    asock_cfg.grp_lock = ssock->param.grp_lock;
+     asock_cfg.async_cnt = ssock->param.async_cnt;
+     asock_cfg.concurrency = ssock->param.concurrency;
+     asock_cfg.whole_data = PJ_TRUE;
+@@ -964,7 +965,7 @@ static pj_bool_t asock_on_accept_complet
+ 	    goto on_return;
+ 
+ 	pj_grp_lock_add_ref(glock);
+-	asock_cfg.grp_lock = ssock->param.grp_lock = glock;
++	ssock->param.grp_lock = glock;
+ 	pj_grp_lock_add_handler(ssock->param.grp_lock, ssock->pool, ssock,
+ 				ssl_on_destroy);
+     }
+--- a/pjlib/src/pj/ssl_sock_ossl.c
++++ b/pjlib/src/pj/ssl_sock_ossl.c
+@@ -327,7 +327,8 @@ static pj_status_t STATUS_FROM_SSL_ERR(c
+ 	ERROR_LOG("STATUS_FROM_SSL_ERR", err, ssock);
+     }
+ 
+-    ssock->last_err = err;
++    if (ssock)
++	ssock->last_err = err;
+     return GET_STATUS_FROM_SSL_ERR(err);
+ }
+ 
+@@ -344,7 +345,8 @@ static pj_status_t STATUS_FROM_SSL_ERR2(
+     /* Dig for more from OpenSSL error queue */
+     SSLLogErrors(action, ret, err, len, ssock);
+ 
+-    ssock->last_err = ssl_err;
++    if (ssock)
++	ssock->last_err = ssl_err;
+     return GET_STATUS_FROM_SSL_ERR(ssl_err);
+ }
+ 
+@@ -587,6 +589,13 @@ static pj_status_t init_openssl(void)
+ 
+     /* Create OpenSSL application data index for SSL socket */
+     sslsock_idx = SSL_get_ex_new_index(0, "SSL socket", NULL, NULL, NULL);
++	if (sslsock_idx == -1) {
++		status = STATUS_FROM_SSL_ERR2("Init", NULL, -1, ERR_get_error(), 0);
++		PJ_LOG(1,(THIS_FILE,
++				  "Fatal error: failed to get application data index for "
++				  "SSL socket"));
++		return status;
++	}
+ 
+     return status;
+ }
+@@ -614,21 +623,36 @@ static int password_cb(char *buf, int nu
+ }
+ 
+ 
+-/* SSL password callback. */
++/* SSL certificate verification result callback.
++ * Note that this callback seems to be always called from library worker
++ * thread, e.g: active socket on_read_complete callback, which should have
++ * already been equipped with race condition avoidance mechanism (should not
++ * be destroyed while callback is being invoked).
++ */
+ static int verify_cb(int preverify_ok, X509_STORE_CTX *x509_ctx)
+ {
+-    pj_ssl_sock_t *ssock;
+-    SSL *ossl_ssl;
++    pj_ssl_sock_t *ssock = NULL;
++    SSL *ossl_ssl = NULL;
+     int err;
+ 
+     /* Get SSL instance */
+     ossl_ssl = X509_STORE_CTX_get_ex_data(x509_ctx, 
+ 				    SSL_get_ex_data_X509_STORE_CTX_idx());
+-    pj_assert(ossl_ssl);
++    if (!ossl_ssl) {
++	PJ_LOG(1,(THIS_FILE,
++		  "SSL verification callback failed to get SSL instance"));
++	goto on_return;
++    }
+ 
+     /* Get SSL socket instance */
+     ssock = SSL_get_ex_data(ossl_ssl, sslsock_idx);
+-    pj_assert(ssock);
++    if (!ssock) {
++	/* SSL socket may have been destroyed */
++	PJ_LOG(1,(THIS_FILE,
++		  "SSL verification callback failed to get SSL socket "
++		  "instance (sslsock_idx=%d).", sslsock_idx));
++	goto on_return;
++    }
+ 
+     /* Store verification status */
+     err = X509_STORE_CTX_get_error(x509_ctx);
+@@ -706,6 +730,7 @@ static int verify_cb(int preverify_ok, X
+     if (PJ_FALSE == ssock->param.verify_peer)
+ 	preverify_ok = 1;
+ 
++on_return:
+     return preverify_ok;
+ }
+ 
+@@ -1213,6 +1238,12 @@ static void ssl_destroy(pj_ssl_sock_t *s
+ static void ssl_reset_sock_state(pj_ssl_sock_t *ssock)
+ {
+     ossl_sock_t *ossock = (ossl_sock_t *)ssock;
++
++    /* Detach from SSL instance */
++    if (ossock->ossl_ssl) {
++	SSL_set_ex_data(ossock->ossl_ssl, sslsock_idx, NULL);
++    }
++
+     /**
+      * Avoid calling SSL_shutdown() if handshake wasn't completed.
+      * OpenSSL 1.0.2f complains if SSL_shutdown() is called during an
diff --git a/libs/pjproject/patches/0120-pjmedia_sdp_attr_get_rtpmap-Strip-param-trailing-whi.patch b/libs/pjproject/patches/0120-pjmedia_sdp_attr_get_rtpmap-Strip-param-trailing-whi.patch
new file mode 100644
index 0000000..a3e668c
--- /dev/null
+++ b/libs/pjproject/patches/0120-pjmedia_sdp_attr_get_rtpmap-Strip-param-trailing-whi.patch
@@ -0,0 +1,27 @@
+From 2ae784030b0d9cf217c3d562af20e4967f19a3dc Mon Sep 17 00:00:00 2001
+From: George Joseph <gjoseph@sangoma.com>
+Date: Tue, 14 Sep 2021 10:47:29 -0600
+Subject: [PATCH] pjmedia_sdp_attr_get_rtpmap: Strip param trailing whitespace
+
+Use pj_scan_get() to parse the param part of rtpmap so
+trailing whitespace is automatically stripped.
+
+Fixes #2827
+---
+ pjmedia/src/pjmedia/sdp.c | 4 ++--
+ 1 file changed, 2 insertions(+), 2 deletions(-)
+
+--- a/pjmedia/src/pjmedia/sdp.c
++++ b/pjmedia/src/pjmedia/sdp.c
+@@ -313,9 +313,9 @@ PJ_DEF(pj_status_t) pjmedia_sdp_attr_get
+ 
+ 	/* Expecting either '/' or EOF */
+ 	if (*scanner.curptr == '/') {
++	    /* Skip the '/' */
+ 	    pj_scan_get_char(&scanner);
+-	    rtpmap->param.ptr = scanner.curptr;
+-	    rtpmap->param.slen = scanner.end - scanner.curptr;
++	    pj_scan_get(&scanner, &cs_token, &rtpmap->param);
+ 	} else {
+ 	    rtpmap->param.slen = 0;
+ 	}
diff --git a/libs/pjproject/patches/170-Fix-build-without-SRTP.patch b/libs/pjproject/patches/170-Fix-build-without-SRTP.patch
deleted file mode 100644
index 8425702..0000000
--- a/libs/pjproject/patches/170-Fix-build-without-SRTP.patch
+++ /dev/null
@@ -1,35 +0,0 @@
-From 42e5ee52e1113d24b7125e29669022c35e87693f Mon Sep 17 00:00:00 2001
-From: Sauw Ming <ming@teluu.com>
-Date: Wed, 26 Sep 2018 00:56:51 +0000
-Subject: [PATCH] Re #2147 (misc): Fixed build error of undefined reference to
- `pjmedia_srtp_enum_crypto' if PJMEDIA_HAS_SRTP is disabled.
-
-Thanks to Alexei Gradinari for the patch
-
-
-
-git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@5891 74dad513-b988-da41-8d7b-12977e46ad98
----
- pjsip/src/pjsua2/endpoint.cpp | 4 +++-
- 1 file changed, 3 insertions(+), 1 deletion(-)
-
---- a/pjsip/src/pjsua2/endpoint.cpp
-+++ b/pjsip/src/pjsua2/endpoint.cpp
-@@ -2145,14 +2145,16 @@ void Endpoint::resetVideoCodecParam(cons
-  */
- StringVector Endpoint::srtpCryptoEnum() throw(Error)
- {
-+    StringVector result;
-+#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
-     unsigned cnt = PJMEDIA_SRTP_MAX_CRYPTOS;
-     pjmedia_srtp_crypto cryptos[PJMEDIA_SRTP_MAX_CRYPTOS];
--    StringVector result;
- 
-     PJSUA2_CHECK_EXPR(pjmedia_srtp_enum_crypto(&cnt, cryptos));
- 
-     for (unsigned i = 0; i < cnt; ++i)
- 	result.push_back(pj2Str(cryptos[i].name));
-+#endif
- 
-     return result;
- }
diff --git a/libs/pjproject/patches/210-outgoing_connected_line_method_update.patch b/libs/pjproject/patches/210-outgoing_connected_line_method_update.patch
deleted file mode 100644
index 201bc3e..0000000
--- a/libs/pjproject/patches/210-outgoing_connected_line_method_update.patch
+++ /dev/null
@@ -1,32 +0,0 @@
---- a/pjsip/src/pjsip-ua/sip_inv.c
-+++ b/pjsip/src/pjsip-ua/sip_inv.c
-@@ -4185,6 +4185,29 @@ static void inv_on_state_calling( pjsip_
- 
- 	    if (tsx->status_code != 100) {
- 
-+		if (inv->role == PJSIP_ROLE_UAC) {
-+		    pjsip_rx_data *rdata = e->body.tsx_state.src.rdata;
-+		    pjsip_allow_hdr *allow = NULL;
-+		    pjsip_msg *msg = rdata->msg_info.msg;
-+
-+		    if (msg) {
-+			allow = (pjsip_allow_hdr*) pjsip_msg_find_hdr(msg, PJSIP_H_ALLOW,
-+				NULL);
-+		    }
-+		    if (allow) {
-+			unsigned i;
-+			const pj_str_t STR_UPDATE = { "UPDATE", 6 };
-+
-+			for (i=0; i<allow->count; ++i) {
-+			    if (pj_stricmp(&allow->values[i], &STR_UPDATE)==0) {
-+				/* UPDATE is present in Allow */
-+				inv->options |= PJSIP_INV_SUPPORT_UPDATE;
-+				break;
-+			    }
-+			}
-+		    }
-+		}
-+
- 		if (dlg->remote.info->tag.slen)
- 		    inv_set_state(inv, PJSIP_INV_STATE_EARLY, e);
- 
diff --git a/libs/pjproject/patches/220-Fixed-2172-Avoid-double-reference-counter-decrements.patch b/libs/pjproject/patches/220-Fixed-2172-Avoid-double-reference-counter-decrements.patch
deleted file mode 100644
index 02634a5..0000000
--- a/libs/pjproject/patches/220-Fixed-2172-Avoid-double-reference-counter-decrements.patch
+++ /dev/null
@@ -1,37 +0,0 @@
-From 1fed39fe1488abd654a5488b5e6ad59b4b973331 Mon Sep 17 00:00:00 2001
-From: nanang <nanang@localhost>
-Date: Tue, 8 Jan 2019 09:07:47 +0000
-Subject: [PATCH 1/5] Fixed #2172: Avoid double reference counter decrements in
- timer in the scenario of race condition between pj_timer_heap_cancel() and
- pj_timer_heap_poll().
-
----
- pjlib/src/pj/timer.c | 17 ++++++++++-------
- 1 file changed, 10 insertions(+), 7 deletions(-)
-
---- a/pjlib/src/pj/timer.c
-+++ b/pjlib/src/pj/timer.c
-@@ -580,13 +580,16 @@ static int cancel_timer(pj_timer_heap_t
- 
-     lock_timer_heap(ht);
-     count = cancel(ht, entry, flags | F_DONT_CALL);
--    if (flags & F_SET_ID) {
--	entry->id = id_val;
--    }
--    if (entry->_grp_lock) {
--	pj_grp_lock_t *grp_lock = entry->_grp_lock;
--	entry->_grp_lock = NULL;
--	pj_grp_lock_dec_ref(grp_lock);
-+    if (count > 0) {
-+	/* Timer entry found & cancelled */
-+	if (flags & F_SET_ID) {
-+	    entry->id = id_val;
-+	}
-+	if (entry->_grp_lock) {
-+	    pj_grp_lock_t *grp_lock = entry->_grp_lock;
-+	    entry->_grp_lock = NULL;
-+	    pj_grp_lock_dec_ref(grp_lock);
-+	}
-     }
-     unlock_timer_heap(ht);
- 
diff --git a/libs/pjproject/patches/230-Re-2176-Removed-pop_freelist-push_freelist-after-rem.patch b/libs/pjproject/patches/230-Re-2176-Removed-pop_freelist-push_freelist-after-rem.patch
deleted file mode 100644
index d0b60e6..0000000
--- a/libs/pjproject/patches/230-Re-2176-Removed-pop_freelist-push_freelist-after-rem.patch
+++ /dev/null
@@ -1,32 +0,0 @@
-From 9f57a5728aaec1949908bf7bbd15768fce74e315 Mon Sep 17 00:00:00 2001
-From: Nanang Izzuddin <nanang@teluu.com>
-Date: Wed, 13 Feb 2019 06:51:09 +0000
-Subject: [PATCH] Re #2176: Removed pop_freelist() + push_freelist() after
- remove_node() as they are not only unnecessary, they cause problem.
-
-git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@5934 74dad513-b988-da41-8d7b-12977e46ad98
----
- pjlib/src/pj/timer.c | 5 +++--
- 1 file changed, 3 insertions(+), 2 deletions(-)
-
---- a/pjlib/src/pj/timer.c
-+++ b/pjlib/src/pj/timer.c
-@@ -633,7 +633,8 @@ PJ_DEF(unsigned) pj_timer_heap_poll( pj_
-     {
- 	pj_timer_entry *node = remove_node(ht, 0);
- 	/* Avoid re-use of this timer until the callback is done. */
--	pj_timer_id_t node_timer_id = pop_freelist(ht);
-+	///Not necessary, even causes problem (see also #2176).
-+	///pj_timer_id_t node_timer_id = pop_freelist(ht);
- 	pj_grp_lock_t *grp_lock;
- 
- 	++count;
-@@ -653,7 +654,7 @@ PJ_DEF(unsigned) pj_timer_heap_poll( pj_
- 
- 	lock_timer_heap(ht);
- 	/* Now, the timer is really free for re-use. */
--	push_freelist(ht, node_timer_id);
-+	///push_freelist(ht, node_timer_id);
-     }
-     if (ht->cur_size && next_delay) {
- 	*next_delay = ht->heap[0]->_timer_value;
diff --git a/net/asterisk-16.x-chan-lantiq/Makefile b/net/asterisk-16.x-chan-lantiq/Makefile
index 5884673..cfc6937 100644
--- a/net/asterisk-16.x-chan-lantiq/Makefile
+++ b/net/asterisk-16.x-chan-lantiq/Makefile
@@ -8,14 +8,14 @@
 include $(TOPDIR)/rules.mk
 
 PKG_NAME:=asterisk16-chan-lantiq
-PKG_VERSION:=20180215
-PKG_RELEASE:=2
+PKG_VERSION:=20210911
+PKG_RELEASE:=1
 
 PKG_SOURCE:=$(PKG_NAME)-$(PKG_VERSION).tar.xz
 PKG_SOURCE_URL:=https://github.com/kochstefan/asterisk_channel_lantiq.git
 PKG_SOURCE_SUBDIR:=$(PKG_NAME)-$(PKG_VERSION)
-PKG_SOURCE_VERSION:=c9d68dd06fcd46ac7985df45f8c2b8833e658f8e
-PKG_MIRROR_HASH:=8666c18b24adf9da3ddf12306fcf0a8b4f56860c256b172bd0ba5c2a7a3ab25e
+PKG_SOURCE_VERSION:=2f029ec8778420538c8151c6aceba0f7b44b07c9
+PKG_MIRROR_HASH:=72ce3cb4a081c6ea39c9814fba03997ef32b48e927469508222f5641fe987c65
 PKG_SOURCE_PROTO:=git
 
 PKG_LICENSE:=GPL-2.0
@@ -43,12 +43,6 @@ define Package/$(PKG_NAME)/conffiles
 /etc/asterisk/lantiq.conf
 endef
 
-define Build/Prepare
-	$(call Build/Prepare/Default)
-	$(INSTALL_DATA) ./files/default.exports \
-		$(PKG_BUILD_DIR)/src/channels/chan_lantiq.exports
-endef
-
 define Build/Compile
 	cd $(PKG_BUILD_DIR)/src/channels && \
 	$(TARGET_CC) -o chan_lantiq.o -c chan_lantiq.c -MD -MT chan_lantiq.o \
diff --git a/net/asterisk-16.x-chan-lantiq/files/default.exports b/net/asterisk-16.x-chan-lantiq/files/default.exports
deleted file mode 100644
index 6d9344d..0000000
--- a/net/asterisk-16.x-chan-lantiq/files/default.exports
+++ /dev/null
@@ -1,8 +0,0 @@
-{
-	global:
-		/* See main/asterisk.exports.in for an explanation why this is
-		 * needed. */
-		_IO_stdin_used;
-	local:
-		*;
-};
diff --git a/net/asterisk-16.x-chan-lantiq/patches/01-Use-ast_malloc-and-ast_free-instead-of-malloc-and-free.patch b/net/asterisk-16.x-chan-lantiq/patches/01-Use-ast_malloc-and-ast_free-instead-of-malloc-and-free.patch
deleted file mode 100644
index f913b18..0000000
--- a/net/asterisk-16.x-chan-lantiq/patches/01-Use-ast_malloc-and-ast_free-instead-of-malloc-and-free.patch
+++ /dev/null
@@ -1,43 +0,0 @@
-commit 30f9a094c1c60e0d68e4ea189f48ecb47aebb485
-Author: arny <arnysch@gmx.net>
-Date:   Thu May 2 20:07:28 2019 +0200
-
-    Use ast_malloc and ast_free instead of malloc and free
-    in order to get rid of build errors with asterisk16 in OpenWrt
-    
-    Signed-off-by: arny <arnysch@gmx.net>
-
-diff --git a/src/channels/chan_lantiq.c b/src/channels/chan_lantiq.c
-index a8fc869..90002ab 100644
---- a/src/channels/chan_lantiq.c
-+++ b/src/channels/chan_lantiq.c
-@@ -563,9 +563,9 @@ lantiq_dev_binary_buffer_create(const char *path, uint8_t **ppBuf, uint32_t *pBu
- 		goto on_exit;
- 	}
- 
--	*ppBuf = malloc(file_stat.st_size);
-+	*ppBuf = ast_malloc(file_stat.st_size);
- 	if (*ppBuf == NULL) {
--		ast_log(LOG_ERROR, "binary file %s memory allocation failed\n", path);
-+		// Message already logged by ast_malloc
- 		goto on_exit;
- 	}
- 
-@@ -583,7 +583,7 @@ on_exit:
- 		fclose(fd);
- 
- 	if (*ppBuf != NULL && status)
--		free(*ppBuf);
-+		ast_free(*ppBuf);
- 
- 	return status;
- }
-@@ -609,7 +609,7 @@ static int32_t lantiq_dev_firmware_download(int32_t fd, const char *path)
- 	}
- 
- 	if (firmware != NULL)
--		free(firmware);
-+		ast_free(firmware);
- 
- 	return 0;
- }
diff --git a/net/asterisk-16.x/Makefile b/net/asterisk-16.x/Makefile
index bd3ad12..14dad42 100644
--- a/net/asterisk-16.x/Makefile
+++ b/net/asterisk-16.x/Makefile
@@ -9,12 +9,12 @@ include $(TOPDIR)/rules.mk
 
 AST_MAJOR_VERSION:=16
 PKG_NAME:=asterisk$(AST_MAJOR_VERSION)
-PKG_VERSION:=$(AST_MAJOR_VERSION).3.0
-PKG_RELEASE:=9
+PKG_VERSION:=$(AST_MAJOR_VERSION).22.0
+PKG_RELEASE:=1
 
 PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
 PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases
-PKG_HASH:=8b22ee7c0c0b5557eff273118703c6fce8b743c12bbeb679ed86b3f197444a8e
+PKG_HASH:=46992482762818e096d92654b9ac96d42fa9505ad4bc8e628a683413793ab26f
 
 PKG_BUILD_DIR:=$(BUILD_DIR)/asterisk-$(PKG_VERSION)
 PKG_BUILD_DEPENDS:=libxml2/host
@@ -43,7 +43,9 @@ MODULES_AVAILABLE:= \
 	app-agent-pool \
 	app-alarmreceiver \
 	app-amd \
+	app-attended-transfer \
 	app-authenticate \
+	app-blind-transfer \
 	app-bridgeaddchan \
 	app-bridgewait \
 	app-celgenuserevent \
@@ -57,6 +59,7 @@ MODULES_AVAILABLE:= \
 	app-directed-pickup \
 	app-directory \
 	app-disa \
+	app-dtmfstore \
 	app-dumpchan \
 	app-exec \
 	app-externalivr \
@@ -67,6 +70,7 @@ MODULES_AVAILABLE:= \
 	app-ices \
 	app-image \
 	app-ivrdemo \
+	app-mf \
 	app-milliwatt \
 	app-minivm \
 	app-mixmonitor \
@@ -80,6 +84,7 @@ MODULES_AVAILABLE:= \
 	app-read \
 	app-readexten \
 	app-record \
+	app-reload \
 	app-saycounted \
 	app-sayunixtime \
 	app-senddtmf \
@@ -99,6 +104,7 @@ MODULES_AVAILABLE:= \
 	app-url \
 	app-userevent \
 	app-verbose \
+	app-waitforcond \
 	app-waitforring \
 	app-waitforsilence \
 	app-waituntil \
@@ -176,6 +182,7 @@ MODULES_AVAILABLE:= \
 	func-enum \
 	func-env \
 	func-extstate \
+	func-frame-drop \
 	func-frame-trace \
 	func-global \
 	func-groupcount \
@@ -192,6 +199,8 @@ MODULES_AVAILABLE:= \
 	func-presencestate \
 	func-rand \
 	func-realtime \
+	func-sayfiles \
+	func-scramble \
 	func-sha1 \
 	func-shell \
 	func-sorcery \
@@ -271,6 +280,7 @@ MODULES_AVAILABLE:= \
 	res-parking \
 	res-phoneprov \
 	res-pjsip-phoneprov \
+	res-pjsip-stir-shaken \
 	res-pjproject \
 	res-pktccops \
 	res-realtime \
@@ -293,10 +303,11 @@ MODULES_AVAILABLE:= \
 	res-stasis-recording \
 	res-stasis-snoop \
 	res-statsd \
+	res-stir-shaken \
 	res-stun-monitor \
 	res-timing-dahdi \
 	res-timing-pthread \
-	res-timing-timerfd \
+	res-tonedetect \
 	res-xmpp \
 	voicemail
 
@@ -485,7 +496,7 @@ AST_CFG_FILES:= \
 AST_EMB_MODULES:=\
 	app_dial app_echo app_macro app_playback \
 	func_callerid func_logic func_strings func_timeout \
-	pbx_config res_crypto
+	pbx_config res_crypto res_timing_timerfd
 
 define Package/$(PKG_NAME)/install
 $(call Package/$(PKG_NAME)/install/lib,$(1),libasteriskssl)
@@ -757,7 +768,9 @@ $(eval $(call BuildAsteriskModule,app-adsiprog,ADSI programming,Asterisk ADSI pr
 $(eval $(call BuildAsteriskModule,app-agent-pool,Call center agent pool,Call center agent pool applications.,,agents.conf,app_agent_pool,,))
 $(eval $(call BuildAsteriskModule,app-alarmreceiver,Alarm receiver,Alarm receiver for Asterisk.,,,app_alarmreceiver,,))
 $(eval $(call BuildAsteriskModule,app-amd,Answering machine detection,Answering Machine Detection application.,,amd.conf,app_amd,,))
+$(eval $(call BuildAsteriskModule,app-attended-transfer,Attended transfer,Queues up an attended transfer to a given extension.,,,app_attended_transfer,,))
 $(eval $(call BuildAsteriskModule,app-authenticate,Authenticate commands,Authentication application.,,,app_authenticate,,))
+$(eval $(call BuildAsteriskModule,app-blind-transfer,Blind transfer,Redirects all channels currently bridged to the caller channel to a specified destination.,,,app_blind_transfer,,))
 $(eval $(call BuildAsteriskModule,app-bridgeaddchan,Bridge add channel,Bridge-add-channel application.,,,app_bridgeaddchan,,))
 $(eval $(call BuildAsteriskModule,app-bridgewait,Holding bridge,Application to place a channel into a holding bridge.,+$(PKG_NAME)-bridge-holding,,app_bridgewait,,))
 $(eval $(call BuildAsteriskModule,app-celgenuserevent,User-defined CEL event,Generate a user defined CEL event.,,,app_celgenuserevent,,))
@@ -771,16 +784,18 @@ $(eval $(call BuildAsteriskModule,app-dictate,Virtual dictation machine,Virtual
 $(eval $(call BuildAsteriskModule,app-directed-pickup,Directed call pickup,Directed call pickup application.,,,app_directed_pickup,,))
 $(eval $(call BuildAsteriskModule,app-directory,Extension directory,Extension directory.,,,app_directory,,))
 $(eval $(call BuildAsteriskModule,app-disa,Direct Inward System Access,Direct Inward System Access application.,,,app_disa,,))
+$(eval $(call BuildAsteriskModule,app-dtmfstore,DTMF storage,Technology independent async DTMF storage.,,,app_dtmfstore,,))
 $(eval $(call BuildAsteriskModule,app-dumpchan,Dump info about channel,Dump info about the calling channel.,,,app_dumpchan,,))
 $(eval $(call BuildAsteriskModule,app-exec,Exec application,Executes dialplan applications.,,,app_exec,,))
 $(eval $(call BuildAsteriskModule,app-externalivr,External IVR interface,External IVR interface application.,,,app_externalivr,,))
 $(eval $(call BuildAsteriskModule,app-festival,Simple festival interface,Simple Festival interface.,,festival.conf,app_festival,,))
 $(eval $(call BuildAsteriskModule,app-flash,Flash channel,Flash channel application.,+$(PKG_NAME)-chan-dahdi,,app_flash,,))
 $(eval $(call BuildAsteriskModule,app-followme,Find-me/follow-me,Find-Me/Follow-Me application.,,followme.conf,app_followme,,))
-$(eval $(call BuildAsteriskModule,app-getcpeid,Get ADSI CPE ID,Get ADSI CPE ID.,,,app_getcpeid,,))
+$(eval $(call BuildAsteriskModule,app-getcpeid,Get ADSI CPE ID,Get ADSI CPE ID.,+$(PKG_NAME)-res-adsi,,app_getcpeid,,))
 $(eval $(call BuildAsteriskModule,app-ices,Encode and stream,Encode and stream via Icecast and IceS.,,,app_ices,,))
 $(eval $(call BuildAsteriskModule,app-image,Image transmission,Image transmission application.,,,app_image,,))
 $(eval $(call BuildAsteriskModule,app-ivrdemo,IVR demo,IVR demo application.,,,app_ivrdemo,,))
+$(eval $(call BuildAsteriskModule,app-mf,MF digits,Send MF digits Application.,,,app_mf,,))
 $(eval $(call BuildAsteriskModule,app-milliwatt,Digital milliwatt [mu-law] test app,Digital milliwatt test application.,,,app_milliwatt,,))
 $(eval $(call BuildAsteriskModule,app-minivm,Minimal voicemail system,A minimal voicemail e-mail system.,,extensions_minivm.conf minivm.conf,app_minivm,,))
 $(eval $(call BuildAsteriskModule,app-mixmonitor,Record a call and mix the audio,Mixed audio monitoring application.,,,app_mixmonitor,,))
@@ -794,6 +809,7 @@ $(eval $(call BuildAsteriskModule,app-queue,True Call Queueing,True call queuein
 $(eval $(call BuildAsteriskModule,app-read,Variable read,Read variable application.,,,app_read,,))
 $(eval $(call BuildAsteriskModule,app-readexten,Extension to variable,Read and evaluate extension validity.,,,app_readexten,,))
 $(eval $(call BuildAsteriskModule,app-record,Record sound file,Trivial record application.,,,app_record,,))
+$(eval $(call BuildAsteriskModule,app-reload,Reload,Reload module[s].,,,app_reload,,))
 $(eval $(call BuildAsteriskModule,app-saycounted,Decline words,Decline words according to channel language.,,,app_saycounted,,))
 $(eval $(call BuildAsteriskModule,app-sayunixtime,Say Unix time,Say time.,,,app_sayunixtime,,))
 $(eval $(call BuildAsteriskModule,app-senddtmf,Send DTMF digits,Send DTMF digits application.,,,app_senddtmf,,))
@@ -813,6 +829,7 @@ $(eval $(call BuildAsteriskModule,app-transfer,Transfers caller to other ext,Tra
 $(eval $(call BuildAsteriskModule,app-url,Send URL,Send URL applications.,,,app_url,,))
 $(eval $(call BuildAsteriskModule,app-userevent,Custom user event,Custom user event application.,,,app_userevent,,))
 $(eval $(call BuildAsteriskModule,app-verbose,Verbose logging,Send verbose output.,,,app_verbose,,))
+$(eval $(call BuildAsteriskModule,app-waitforcond,Wait for condition,Wait until condition is true.,,,app_waitforcond,,))
 $(eval $(call BuildAsteriskModule,app-waitforring,Wait for first ring,Waits until first ring after time.,,,app_waitforring,,))
 $(eval $(call BuildAsteriskModule,app-waitforsilence,Wait for silence/noise,Wait for silence/noise.,,,app_waitforsilence,,))
 $(eval $(call BuildAsteriskModule,app-waituntil,Sleep,Wait until specified time.,,,app_waituntil,,))
@@ -834,14 +851,14 @@ $(eval $(call BuildAsteriskModule,chan-alsa,ALSA channel,ALSA console channel dr
 $(eval $(call BuildAsteriskModule,chan-bridge-media,Bridge media channel driver,Bridge media channel driver.,,,chan_bridge_media,,))
 $(eval $(call BuildAsteriskModule,chan-console,Console channel driver,Console channel driver.,+portaudio,console.conf,chan_console,,))
 $(eval $(call BuildAsteriskModule,chan-dahdi,DAHDI channel,DAHDI telephony.,+dahdi-tools-libtonezone +kmod-dahdi +libpri @!aarch64,chan_dahdi.conf,chan_dahdi,,))
-$(eval $(call BuildAsteriskModule,chan-iax2,IAX2 channel,Inter Asterisk eXchange.,+$(PKG_NAME)-res-timing-timerfd,iax.conf iaxprov.conf,chan_iax2,,))
+$(eval $(call BuildAsteriskModule,chan-iax2,IAX2 channel,Inter Asterisk eXchange.,,iax.conf iaxprov.conf,chan_iax2,,))
 $(eval $(call BuildAsteriskModule,chan-mgcp,MGCP,Media Gateway Control Protocol.,,mgcp.conf,chan_mgcp,,))
 $(eval $(call BuildAsteriskModule,chan-mobile,Bluetooth channel,Bluetooth mobile device channel driver.,+bluez-libs,chan_mobile.conf,chan_mobile,,))
 $(eval $(call BuildAsteriskModule,chan-motif,Jingle channel,Motif Jingle channel driver.,+$(PKG_NAME)-res-xmpp,motif.conf,chan_motif,,))
 $(eval $(call BuildAsteriskModule,chan-ooh323,H.323 channel,Objective Systems H.323 channel.,,ooh323.conf,chan_ooh323,,))
 $(eval $(call BuildAsteriskModule,chan-oss,OSS channel,OSS console channel driver.,,oss.conf,chan_oss,,))
 $(eval $(call BuildAsteriskModule,chan-phone,Linux telephony API,Linux telephony API support.,,phone.conf,chan_phone,,))
-$(eval $(call BuildAsteriskModule,chan-rtp,RTP media channel,RTP media channel.,,,chan_rtp,,))
+$(eval $(call BuildAsteriskModule,chan-rtp,RTP media channel,RTP media channel.,+$(PKG_NAME)-res-rtp-multicast,,chan_rtp,,))
 $(eval $(call BuildAsteriskModule,chan-sip,SIP channel,Session Initiation Protocol.,+$(PKG_NAME)-app-confbridge,sip.conf sip_notify.conf,chan_sip,,))
 $(eval $(call BuildAsteriskModule,chan-skinny,Skinny channel,Skinny Client Control Protocol.,,skinny.conf,chan_skinny,,))
 $(eval $(call BuildAsteriskModule,chan-unistim,Unistim channel,UNISTIM protocol.,,unistim.conf,chan_unistim,,))
@@ -890,6 +907,7 @@ $(eval $(call BuildAsteriskModule,func-dialplan,Dialplan context/extension/prior
 $(eval $(call BuildAsteriskModule,func-enum,ENUM,ENUM related dialplan functions.,,enum.conf,func_enum,,))
 $(eval $(call BuildAsteriskModule,func-env,Environment functions,Environment/filesystem dialplan functions.,,,func_env,,))
 $(eval $(call BuildAsteriskModule,func-extstate,Hinted extension state,Gets the state of an extension in the dialplan.,,,func_extstate,,))
+$(eval $(call BuildAsteriskModule,func-frame-drop,Frame drop,Function to drop frames on a channel.,,,func_frame_drop,,))
 $(eval $(call BuildAsteriskModule,func-frame-trace,Frame trace for internal ast_frame debugging,Frame trace for internal ast_frame debugging.,,,func_frame_trace,,))
 $(eval $(call BuildAsteriskModule,func-global,Global variable,Variable dialplan functions.,,,func_global,,))
 $(eval $(call BuildAsteriskModule,func-groupcount,Group count,Channel group dialplan functions.,,,func_groupcount,,))
@@ -906,6 +924,8 @@ $(eval $(call BuildAsteriskModule,func-pitchshift,Audio effects dialplan functio
 $(eval $(call BuildAsteriskModule,func-presencestate,Hinted presence state,Gets or sets a presence state in the dialplan.,,,func_presencestate,,))
 $(eval $(call BuildAsteriskModule,func-rand,RAND dialplan function,Random number dialplan function.,,,func_rand,,))
 $(eval $(call BuildAsteriskModule,func-realtime,REALTIME dialplan function,Read/write/store/destroy values from a realtime repository.,,,func_realtime,,))
+$(eval $(call BuildAsteriskModule,func-sayfiles,Say files,Say application files.,,,func_sayfiles,,))
+$(eval $(call BuildAsteriskModule,func-scramble,Scramble,Frequency inverting voice scrambler.,,,func_scramble,,))
 $(eval $(call BuildAsteriskModule,func-sha1,SHA-1 computation dialplan function,SHA-1 computation dialplan function.,,,func_sha1,,))
 $(eval $(call BuildAsteriskModule,func-shell,Shell,Collects the output generated by a command executed by the system shell.,,,func_shell,,))
 $(eval $(call BuildAsteriskModule,func-sorcery,Get a field from a sorcery object,Get a field from a sorcery object.,,,func_sorcery,,))
@@ -960,7 +980,7 @@ $(eval $(call BuildAsteriskModule,res-hep,HEPv3 API,HEPv3 API.,,hep.conf,res_hep
 $(eval $(call BuildAsteriskModule,res-hep-pjsip,PJSIP HEPv3 Logger,PJSIP HEPv3 logger.,+$(PKG_NAME)-res-hep +$(PKG_NAME)-pjsip,,res_hep_pjsip,,))
 $(eval $(call BuildAsteriskModule,res-hep-rtcp,RTCP HEPv3 Logger,RTCP HEPv3 logger.,+$(PKG_NAME)-res-hep,,res_hep_rtcp,,))
 $(eval $(call BuildAsteriskModule,res-fax-spandsp,Spandsp T.38 and G.711,Spandsp G.711 and T.38 FAX technologies.,+$(PKG_NAME)-res-fax +libspandsp +libtiff,,res_fax_spandsp,,))
-$(eval $(call BuildAsteriskModule,res-fax,FAX modules,Generic FAX applications.,+$(PKG_NAME)-res-timing-pthread,res_fax.conf,res_fax,,))
+$(eval $(call BuildAsteriskModule,res-fax,FAX modules,Generic FAX applications.,,res_fax.conf,res_fax,,))
 $(eval $(call BuildAsteriskModule,res-format-attr-celt,CELT format attribute module,CELT format attribute module.,,,res_format_attr_celt,,))
 $(eval $(call BuildAsteriskModule,res-format-attr-g729,G.729 format attribute module,G.729 format attribute module.,,,res_format_attr_g729,,))
 $(eval $(call BuildAsteriskModule,res-format-attr-h263,H.263 format attribute module,H.263 format attribute module.,,,res_format_attr_h263,,))
@@ -984,8 +1004,9 @@ $(eval $(call BuildAsteriskModule,res-mwi-external,Core external MWI resource,Co
 $(eval $(call BuildAsteriskModule,res-mwi-external-ami,AMI for external MWI,AMI support for external MWI.,+$(PKG_NAME)-res-mwi-external,,res_mwi_external_ami,,))
 $(eval $(call BuildAsteriskModule,res-parking,Phone Parking,Call parking resource.,+$(PKG_NAME)-bridge-holding,res_parking.conf,res_parking,,))
 $(eval $(call BuildAsteriskModule,res-phoneprov,Phone Provisioning,HTTP phone provisioning.,,phoneprov.conf,res_phoneprov,,))
+$(eval $(call BuildAsteriskModule,res-pjsip-stir-shaken,PJSIP STIR/SHAKEN resource,PJSIP STIR/SHAKEN resource module.,+$(PKG_NAME)-pjsip +$(PKG_NAME)-res-stir-shaken,,res_pjsip_stir_shaken,,))
 $(eval $(call BuildAsteriskModule,res-pjsip-phoneprov,PJSIP Phone Provisioning,PJSIP phone provisioning.,+$(PKG_NAME)-pjsip +$(PKG_NAME)-res-phoneprov,,res_pjsip_phoneprov_provider,,))
-$(eval $(call BuildAsteriskModule,res-pjproject,Bridge PJPROJECT to Asterisk logging,PJProject log and utility support.,+libpj +libpjlib-util +libpjmedia +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +libpjsua +libpjsua2 +libsrtp2,pjproject.conf,res_pjproject,,))
+$(eval $(call BuildAsteriskModule,res-pjproject,Bridge PJPROJECT to Asterisk logging,PJProject log and utility support.,+libpj +libpjlib-util +libpjmedia +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +libpjsua +libpjsua2 +libsrtp2 +$(PKG_NAME)-res-sorcery,pjproject.conf,res_pjproject,,))
 $(eval $(call BuildAsteriskModule,res-pktccops,PktcCOPS manager for MGCP,PktcCOPS manager for MGCP.,,res_pktccops.conf,res_pktccops,,))
 $(eval $(call BuildAsteriskModule,res-realtime,RealTime CLI,Realtime data lookup/rewrite.,,,res_realtime,,))
 $(eval $(call BuildAsteriskModule,res-remb-modifier,REMB modifier,REMB modifier module.,,,res_remb_modifier,,))
@@ -1007,10 +1028,11 @@ $(eval $(call BuildAsteriskModule,res-stasis-playback,Stasis application playbac
 $(eval $(call BuildAsteriskModule,res-stasis-recording,Stasis application recording,Stasis application recording support.,+$(PKG_NAME)-res-stasis,,res_stasis_recording,,))
 $(eval $(call BuildAsteriskModule,res-stasis-snoop,Stasis application snoop,Stasis application snoop support.,+$(PKG_NAME)-res-stasis-recording,,res_stasis_snoop,,))
 $(eval $(call BuildAsteriskModule,res-statsd,statsd client,Statsd client support.,,statsd.conf,res_statsd,,))
+$(eval $(call BuildAsteriskModule,res-stir-shaken,STIR/SHAKEN resource module,STIR/SHAKEN resource module.,+$(PKG_NAME)-curl,stir_shaken.conf,res_stir_shaken,,))
 $(eval $(call BuildAsteriskModule,res-stun-monitor,STUN monitoring,STUN network monitor.,,res_stun_monitor.conf,res_stun_monitor,,))
 $(eval $(call BuildAsteriskModule,res-timing-dahdi,DAHDI Timing Interface,DAHDI timing interface.,+$(PKG_NAME)-chan-dahdi,,res_timing_dahdi,,))
 $(eval $(call BuildAsteriskModule,res-timing-pthread,pthread Timing Interface,pthread timing interface.,,,res_timing_pthread,,))
-$(eval $(call BuildAsteriskModule,res-timing-timerfd,Timerfd Timing Interface,Timerfd timing interface.,,,res_timing_timerfd,,))
+$(eval $(call BuildAsteriskModule,res-tonedetect,Tone detection,Tone detection module.,,,res_tonedetect,,))
 $(eval $(call BuildAsteriskModule,res-xmpp,XMPP client and component module,Asterisk XMPP interface.,+libiksemel +libopenssl,xmpp.conf,res_xmpp,,))
 $(eval $(call BuildAsteriskModule,voicemail,Voicemail,Voicemail modules.,+$(PKG_NAME)-res-adsi +$(PKG_NAME)-res-smdi,voicemail.conf,app_voicemail,vm-*,))
 
diff --git a/net/asterisk-16.x/patches/001-disable-semaphores-on-uclibc-otherwise-allow.patch b/net/asterisk-16.x/patches/001-disable-semaphores-on-uclibc-otherwise-allow.patch
index 7485608..c295cc6 100644
--- a/net/asterisk-16.x/patches/001-disable-semaphores-on-uclibc-otherwise-allow.patch
+++ b/net/asterisk-16.x/patches/001-disable-semaphores-on-uclibc-otherwise-allow.patch
@@ -1,6 +1,6 @@
 --- a/configure.ac
 +++ b/configure.ac
-@@ -1018,15 +1018,18 @@ AC_LINK_IFELSE(
+@@ -1033,15 +1033,18 @@ AC_LINK_IFELSE(
  
  # Some platforms define sem_init(), but only support sem_open(). joyous.
  AC_MSG_CHECKING(for working unnamed semaphores)
diff --git a/net/asterisk-16.x/patches/002-configure-fix-detection-of-re-entrant-resolver-funct.patch b/net/asterisk-16.x/patches/002-configure-fix-detection-of-re-entrant-resolver-funct.patch
index e35ef4c..c7b91e7 100644
--- a/net/asterisk-16.x/patches/002-configure-fix-detection-of-re-entrant-resolver-funct.patch
+++ b/net/asterisk-16.x/patches/002-configure-fix-detection-of-re-entrant-resolver-funct.patch
@@ -18,7 +18,7 @@ Signed-off-by: Bernd Kuhls <bernd.kuhls@t-online.de>
 
 --- a/configure.ac
 +++ b/configure.ac
-@@ -1412,7 +1412,11 @@ AC_LINK_IFELSE(
+@@ -1427,7 +1427,11 @@ AC_LINK_IFELSE(
  			#include <arpa/nameser.h>
  			#endif
  			#include <resolv.h>],
diff --git a/net/asterisk-16.x/patches/056-fix-check_expr2-build.patch b/net/asterisk-16.x/patches/056-fix-check_expr2-build.patch
deleted file mode 100644
index 63a1267..0000000
--- a/net/asterisk-16.x/patches/056-fix-check_expr2-build.patch
+++ /dev/null
@@ -1,38 +0,0 @@
-From 06e8d5ad8e4728a716bf357c8d7f70367ae10280 Mon Sep 17 00:00:00 2001
-From: Sebastian Kemper <sebastian_ml@gmx.net>
-Date: Sun, 12 Jan 2020 12:37:46 +0100
-Subject: [PATCH] check_expr2: fix cross-compile/hardening issues
-
-When building check_expr2 with ASLR PIE hardening enabled the linker
-fails. This is resolved by adding the regular compiler flags when
-building the object files from ast_expr2f.c and ast_expr2.c.
-
-Note: The STANDALONE define is removed because it is already defined in
-_ASTCFLAGS. YY_NO_INPUT is defined so that the compile survives
-'--enable-dev-mode'.
-
-ASTERISK-28685 #close
-
-Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
-Change-Id: If435b7db9f9ad8266245bda51c81c220f9658915
-Taken just Makefile changes from commit: 06e8d5ad8e4728a716bf357c8d7f70367ae10280
----
---- a/utils/Makefile
-+++ b/utils/Makefile
-@@ -180,14 +180,13 @@ conf2ael: conf2ael.o ast_expr2f.o ast_ex
- 
- check_expr2: $(ASTTOPDIR)/main/ast_expr2f.c $(ASTTOPDIR)/main/ast_expr2.c $(ASTTOPDIR)/main/ast_expr2.h astmm.o
- 	$(ECHO_PREFIX) echo "   [CC] ast_expr2f.c -> ast_expr2fz.o"
--	$(CC) -g -c -I$(ASTTOPDIR)/include -DSTANDALONE $(ASTTOPDIR)/main/ast_expr2f.c -o ast_expr2fz.o
-+	$(CC) -g -c -I$(ASTTOPDIR)/include $(_ASTCFLAGS) $(ASTTOPDIR)/main/ast_expr2f.c -o ast_expr2fz.o
- 	$(ECHO_PREFIX) echo "   [CC] ast_expr2.c -> ast_expr2z.o"
--	$(CC) -g -c -I$(ASTTOPDIR)/include -DSTANDALONE2 $(ASTTOPDIR)/main/ast_expr2.c -o ast_expr2z.o
-+	$(CC) -g -c -I$(ASTTOPDIR)/include $(_ASTCFLAGS) -DSTANDALONE2 $(ASTTOPDIR)/main/ast_expr2.c -o ast_expr2z.o
- 	$(ECHO_PREFIX) echo "   [LD] ast_expr2fz.o ast_expr2z.o  -> check_expr2"
- 	$(CC) -g -o check_expr2 ast_expr2fz.o ast_expr2z.o astmm.o -lm $(_ASTLDFLAGS)
- 	$(ECHO_PREFIX) echo "   [RM] ast_expr2fz.o ast_expr2z.o"
- 	rm ast_expr2z.o ast_expr2fz.o
--	./check_expr2 expr2.testinput
- 
- smsq: smsq.o strcompat.o
- smsq: LIBS+=$(POPT_LIB)
diff --git a/net/asterisk-16.x/patches/100-build-reproducibly.patch b/net/asterisk-16.x/patches/100-build-reproducibly.patch
index e7116ff..b4f017c 100644
--- a/net/asterisk-16.x/patches/100-build-reproducibly.patch
+++ b/net/asterisk-16.x/patches/100-build-reproducibly.patch
@@ -17,12 +17,12 @@
   * build.h
 --- a/Makefile
 +++ b/Makefile
-@@ -484,7 +484,7 @@ doc/core-en_US.xml: makeopts .lastclean
+@@ -489,7 +489,7 @@ doc/core-en_US.xml: makeopts .lastclean
  	@echo "<docs xmlns:xi=\"http://www.w3.org/2001/XInclude\">" >> $@
  	@for x in $(MOD_SUBDIRS); do \
  		printf "$$x " ; \
 -		for i in `find $$x -name '*.c'`; do \
 +		for i in `find $$x -name '*.c' | LC_ALL=C sort`; do \
- 			$(AWK) -f build_tools/get_documentation $$i >> $@ ; \
- 		done ; \
- 	done
+ 			MODULEINFO=$$($(AWK) -f build_tools/get_moduleinfo $$i) ; \
+ 			if [ -n "$$MODULEINFO" ] ; \
+ 			then \
diff --git a/net/asterisk-16.x/patches/120-loader.patch b/net/asterisk-16.x/patches/120-loader.patch
deleted file mode 100644
index c9dd9f2..0000000
--- a/net/asterisk-16.x/patches/120-loader.patch
+++ /dev/null
@@ -1,247 +0,0 @@
-commit 02fda2b478f98cf3b8a1df76f772bf0be73bddd5
-Author: Sebastian Kemper <sebastian_ml@gmx.net>
-Date:   Tue Apr 2 22:49:52 2019 +0200
-
-    loader: support for permanent dlopen()
-    
-    Asterisk assumes that dlopen() will always run the constructor of a
-    shared library and every dlclose() will run its destructor. But dlopen()
-    may be permanent, meaning the constructor will only be run once, as is
-    the case with musl libc.
-    
-    With a permanent dlopen() the Asterisk module loader does not work
-    correctly, because it's expectations regarding when the constructors and
-    destructors are run are not met. In fact a segmentation fault will occur
-    when the first module is "re-opened" that has AST_MODFLAG_GLOBAL_SYMBOLS
-    set (the dlopen() does not call the constructor, resource_being_loaded
-    is not set to NULL, then strlen is called with NULL instead of a string,
-    see issue ASTERISK-28319).
-    
-    This commit adds code to the loader that will manually run the
-    constructors/destructors of the (non-builtin) modules where needed. To
-    achieve this a new ao2 container (linked list) is started and filled
-    with objects that contain the names of the modules and the pointers to
-    their respective info structs.
-    
-    This behavior can be activated when configuring Asterisk
-    (--enable-permanent-dlopen). By default this is disabled, of course.
-    
-    ASTERISK-28319 #close
-    
-    Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
-    Change-Id: I86693a0ecf25d5ba81c73773a03df4abc3426875
-
---- a/configure.ac
-+++ b/configure.ac
-@@ -727,6 +727,20 @@ if test "${DISABLE_XMLDOC}" != "yes"; th
- 
- fi
- 
-+AC_ARG_ENABLE([permanent-dlopen],
-+	[AS_HELP_STRING([--enable-permanent-dlopen],
-+		[Enable when your libc has a permanent dlopen like musl])],
-+	[case "${enableval}" in
-+		y|ye|yes) PERMANENT_DLOPEN=yes ;;
-+		n|no)  PERMANENT_DLOPEN=no ;;
-+		*) AC_MSG_ERROR(bad value ${enableval} for --enable-permanent-dlopen)  ;;
-+	esac], [PERMANENT_DLOPEN=no])
-+
-+AC_SUBST([PERMANENT_DLOPEN])
-+if test "${PERMANENT_DLOPEN}" == "yes"; then
-+	AC_DEFINE([HAVE_PERMANENT_DLOPEN], 1, [Define to support libc with permanent dlopen.])
-+fi
-+
- # some embedded systems omit internationalization (locale) support
- AC_CHECK_HEADERS([xlocale.h])
- 
---- a/main/loader.c
-+++ b/main/loader.c
-@@ -153,6 +153,117 @@ static unsigned int loader_ready;
- static struct ast_vector_string startup_errors;
- static struct ast_str *startup_error_builder;
- 
-+#if defined(HAVE_PERMANENT_DLOPEN)
-+#define FIRST_DLOPEN 999
-+
-+struct ao2_container *info_list = NULL;
-+
-+struct info_list_obj {
-+	const struct ast_module_info *info;
-+	int dlopened;
-+	char name[0];
-+};
-+
-+static struct info_list_obj *info_list_obj_alloc(const char *name,
-+	const struct ast_module_info *info)
-+{
-+	struct info_list_obj *new_entry;
-+
-+	new_entry = ao2_alloc(sizeof(*new_entry) + strlen(name) + 1, NULL);
-+
-+	if (!new_entry) {
-+		return NULL;
-+	}
-+
-+	strcpy(new_entry->name, name); /* SAFE */
-+	new_entry->info = info;
-+	new_entry->dlopened = FIRST_DLOPEN;
-+
-+	return new_entry;
-+}
-+
-+AO2_STRING_FIELD_CMP_FN(info_list_obj, name)
-+
-+static char *get_name_from_resource(const char *resource)
-+{
-+	int len;
-+	const char *last_three;
-+	char *mod_name;
-+
-+	if (!resource) {
-+		return NULL;
-+	}
-+
-+	len = strlen(resource);
-+	if (len > 3) {
-+		last_three = &resource[len-3];
-+		if (!strcasecmp(last_three, ".so")) {
-+			mod_name = ast_calloc(1, len - 2);
-+			if (mod_name) {
-+				ast_copy_string(mod_name, resource, len - 2);
-+				return mod_name;
-+			} else {
-+				/* Unable to allocate memory. */
-+				return NULL;
-+			}
-+		}
-+	}
-+
-+	/* Resource is the name - happens when manually unloading a module. */
-+	mod_name = ast_calloc(1, len + 1);
-+	if (mod_name) {
-+		ast_copy_string(mod_name, resource, len + 1);
-+		return mod_name;
-+	}
-+
-+	/* Unable to allocate memory. */
-+	return NULL;
-+}
-+
-+static void manual_mod_reg(const void *lib, const char *resource)
-+{
-+	struct info_list_obj *obj_tmp;
-+	char *mod_name;
-+
-+	if (lib) {
-+		mod_name = get_name_from_resource(resource);
-+		if (mod_name) {
-+			obj_tmp = ao2_find(info_list, mod_name, OBJ_SEARCH_KEY);
-+			if (obj_tmp) {
-+				if (obj_tmp->dlopened == FIRST_DLOPEN) {
-+					obj_tmp->dlopened = 1;
-+				} else {
-+					ast_module_register(obj_tmp->info);
-+				}
-+				ao2_ref(obj_tmp, -1);
-+			}
-+			ast_free(mod_name);
-+		}
-+	}
-+}
-+
-+static void manual_mod_unreg(const char *resource)
-+{
-+	struct info_list_obj *obj_tmp;
-+	char *mod_name;
-+
-+	/* When Asterisk shuts down the destructor is called automatically. */
-+	if (ast_shutdown_final()) {
-+		return;
-+	}
-+
-+	mod_name = get_name_from_resource(resource);
-+	if (mod_name) {
-+		obj_tmp = ao2_find(info_list, mod_name, OBJ_SEARCH_KEY);
-+		if (obj_tmp) {
-+			ast_module_unregister(obj_tmp->info);
-+			ao2_ref(obj_tmp, -1);
-+		}
-+		ast_free(mod_name);
-+	}
-+}
-+#endif
-+
- static __attribute__((format(printf, 1, 2))) void module_load_error(const char *fmt, ...)
- {
- 	char *copy = NULL;
-@@ -597,6 +708,23 @@ void ast_module_register(const struct as
- 
- 	/* give the module a copy of its own handle, for later use in registrations and the like */
- 	*((struct ast_module **) &(info->self)) = mod;
-+
-+#if defined(HAVE_PERMANENT_DLOPEN)
-+	if (mod->flags.builtin != 1) {
-+		struct info_list_obj *obj_tmp = ao2_find(info_list, info->name,
-+			OBJ_SEARCH_KEY);
-+
-+		if (!obj_tmp) {
-+			obj_tmp = info_list_obj_alloc(info->name, info);
-+			if (obj_tmp) {
-+				ao2_link(info_list, obj_tmp);
-+				ao2_ref(obj_tmp, -1);
-+			}
-+		} else {
-+			ao2_ref(obj_tmp, -1);
-+		}
-+	}
-+#endif
- }
- 
- static int module_post_register(struct ast_module *mod)
-@@ -843,6 +971,10 @@ static void logged_dlclose(const char *n
- 		error = dlerror();
- 		ast_log(AST_LOG_ERROR, "Failure in dlclose for module '%s': %s\n",
- 			S_OR(name, "unknown"), S_OR(error, "Unknown error"));
-+#if defined(HAVE_PERMANENT_DLOPEN)
-+	} else {
-+		manual_mod_unreg(name);
-+#endif
- 	}
- }
- 
-@@ -949,6 +1081,9 @@ static struct ast_module *load_dlopen(co
- 
- 	resource_being_loaded = mod;
- 	mod->lib = dlopen(filename, flags);
-+#if defined(HAVE_PERMANENT_DLOPEN)
-+	manual_mod_reg(mod->lib, mod->resource);
-+#endif
- 	if (resource_being_loaded) {
- 		struct ast_str *list;
- 		int c = 0;
-@@ -968,6 +1103,9 @@ static struct ast_module *load_dlopen(co
- 
- 		resource_being_loaded = mod;
- 		mod->lib = dlopen(filename, RTLD_LAZY | RTLD_LOCAL);
-+#if defined(HAVE_PERMANENT_DLOPEN)
-+		manual_mod_reg(mod->lib, mod->resource);
-+#endif
- 		if (resource_being_loaded) {
- 			resource_being_loaded = NULL;
- 
-@@ -2206,6 +2344,15 @@ int load_modules(void)
- 
- 	ast_verb(1, "Asterisk Dynamic Loader Starting:\n");
- 
-+#if defined(HAVE_PERMANENT_DLOPEN)
-+	info_list = ao2_container_alloc_list(AO2_ALLOC_OPT_LOCK_NOLOCK, 0, NULL,
-+		info_list_obj_cmp_fn); /* must not be cleaned at shutdown */
-+	if (!info_list) {
-+		fprintf(stderr, "Module info list allocation failure.\n");
-+		return 1;
-+	}
-+#endif
-+
- 	AST_LIST_HEAD_INIT_NOLOCK(&load_order);
- 	AST_DLLIST_LOCK(&module_list);
- 
diff --git a/net/asterisk-16.x/patches/130-eventfd.patch b/net/asterisk-16.x/patches/130-eventfd.patch
index c783a52..de4441b 100644
--- a/net/asterisk-16.x/patches/130-eventfd.patch
+++ b/net/asterisk-16.x/patches/130-eventfd.patch
@@ -1,6 +1,6 @@
 --- a/configure.ac
 +++ b/configure.ac
-@@ -1205,7 +1205,7 @@ if test "${ac_cv_have_variable_fdset}x"
+@@ -1206,7 +1206,7 @@ if test "${ac_cv_have_variable_fdset}x"
  fi
  
  AC_MSG_CHECKING([if we have usable eventfd support])
diff --git a/net/asterisk-16.x/patches/140-AST-2019-002-16.diff b/net/asterisk-16.x/patches/140-AST-2019-002-16.diff
deleted file mode 100644
index 635d837..0000000
--- a/net/asterisk-16.x/patches/140-AST-2019-002-16.diff
+++ /dev/null
@@ -1,40 +0,0 @@
-From 785bf3a755e47d92caef110e6040295764d08127 Mon Sep 17 00:00:00 2001
-From: George Joseph <gjoseph@digium.com>
-Date: Wed, 12 Jun 2019 12:03:04 -0600
-Subject: [PATCH] res_pjsip_messaging:  Check for body in in-dialog message
-
-We now check that a body exists and it has a length > 0 before
-attempting to process it.
-
-ASTERISK-28447
-Reported-by: Gil Richard
-
-Change-Id: Ic469544b22ab848734636588d4c93426cc6f4b1f
----
- res/res_pjsip_messaging.c | 9 ++++++---
- 1 file changed, 6 insertions(+), 3 deletions(-)
-
-diff --git a/res/res_pjsip_messaging.c b/res/res_pjsip_messaging.c
-index 0e10a8f047..930cf84a53 100644
---- a/res/res_pjsip_messaging.c
-+++ b/res/res_pjsip_messaging.c
-@@ -90,10 +90,13 @@ static enum pjsip_status_code check_content_type_in_dialog(const pjsip_rx_data *
- 	static const pj_str_t text = { "text", 4};
- 	static const pj_str_t application = { "application", 11};
- 
-+	if (!(rdata->msg_info.msg->body && rdata->msg_info.msg->body->len > 0)) {
-+		return res;
-+	}
-+
- 	/* We'll accept any text/ or application/ content type */
--	if (rdata->msg_info.msg->body && rdata->msg_info.msg->body->len
--		&& (pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &text) == 0
--			|| pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &application) == 0)) {
-+	if (pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &text) == 0
-+			|| pj_stricmp(&rdata->msg_info.msg->body->content_type.type, &application) == 0) {
- 		res = PJSIP_SC_OK;
- 	} else if (rdata->msg_info.ctype
- 		&& (pj_stricmp(&rdata->msg_info.ctype->media.type, &text) == 0
--- 
-2.21.0
-
diff --git a/net/asterisk-16.x/patches/150-AST-2019-003-16.diff b/net/asterisk-16.x/patches/150-AST-2019-003-16.diff
deleted file mode 100644
index 90b9d5d..0000000
--- a/net/asterisk-16.x/patches/150-AST-2019-003-16.diff
+++ /dev/null
@@ -1,39 +0,0 @@
-From 1e4df0215af4f192ed06a7fc7589c799f1ec6091 Mon Sep 17 00:00:00 2001
-From: Francesco Castellano <francesco.castellano@messagenet.it>
-Date: Fri, 28 Jun 2019 18:15:31 +0200
-Subject: [PATCH] chan_sip: Handle invalid SDP answer to T.38 re-invite
-
-The chan_sip module performs a T.38 re-invite using a single media
-stream of udptl, and expects the SDP answer to be the same.
-
-If an SDP answer is received instead that contains an additional
-media stream with no joint codec a crash will occur as the code
-assumes that at least one joint codec will exist in this
-scenario.
-
-This change removes this assumption.
-
-ASTERISK-28465
-
-Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87
----
-
-diff --git a/channels/chan_sip.c b/channels/chan_sip.c
-index 898b646..a609ff8 100644
---- a/channels/chan_sip.c
-+++ b/channels/chan_sip.c
-@@ -10965,7 +10965,13 @@
- 			    ast_rtp_lookup_mime_multiple2(s3, NULL, newnoncodeccapability, 0, 0));
- 	}
- 
--	if (portno != -1 || vportno != -1 || tportno != -1) {
-+	/* When UDPTL is negotiated it is expected that there are no compatible codecs as audio or
-+	 * video is not being transported, thus we continue in this function further up if that is
-+	 * the case. If we receive an SDP answer containing both a UDPTL stream and another media
-+	 * stream however we need to check again to ensure that there is at least one joint codec
-+	 * instead of assuming there is one.
-+	 */
-+	if ((portno != -1 || vportno != -1 || tportno != -1) && ast_format_cap_count(newjointcapability)) {
- 		/* We are now ready to change the sip session and RTP structures with the offered codecs, since
- 		   they are acceptable */
- 		unsigned int framing;
diff --git a/net/asterisk-16.x/patches/160-AST-2019-004-16.patch b/net/asterisk-16.x/patches/160-AST-2019-004-16.patch
deleted file mode 100644
index b97b911..0000000
--- a/net/asterisk-16.x/patches/160-AST-2019-004-16.patch
+++ /dev/null
@@ -1,171 +0,0 @@
-From 69ed619a6d9b64a297d3099b6455756912e21d0b Mon Sep 17 00:00:00 2001
-From: Kevin Harwell <kharwell@digium.com>
-Date: Tue, 20 Aug 2019 15:05:45 -0500
-Subject: [PATCH] AST-2019-004 - res_pjsip_t38.c: Add NULL checks before using session media
-
-After receiving a 200 OK with a declined stream in response to a T.38
-initiated re-invite Asterisk would crash when attempting to dereference
-a NULL session media object.
-
-This patch checks to make sure the session media object is not NULL before
-attempting to use it.
-
-ASTERISK-28495
-patches:
-  ast-2019-004.patch submitted by Alexei Gradinari (license 5691)
-
-Change-Id: I168f45f4da29cfe739acf87e597baa2aae7aa572
----
-
-diff --git a/res/res_pjsip_t38.c b/res/res_pjsip_t38.c
-index 11804e2..e5c6090 100644
---- a/res/res_pjsip_t38.c
-+++ b/res/res_pjsip_t38.c
-@@ -203,7 +203,6 @@
- {
- 	RAII_VAR(struct ast_sip_session *, session, obj, ao2_cleanup);
- 	RAII_VAR(struct ast_datastore *, datastore, ast_sip_session_get_datastore(session, "t38"), ao2_cleanup);
--	struct ast_sip_session_media *session_media;
- 
- 	if (!datastore) {
- 		return 0;
-@@ -212,8 +211,7 @@
- 	ast_debug(2, "Automatically rejecting T.38 request on channel '%s'\n",
- 		session->channel ? ast_channel_name(session->channel) : "<gone>");
- 
--	session_media = session->pending_media_state->default_session[AST_MEDIA_TYPE_IMAGE];
--	t38_change_state(session, session_media, datastore->data, T38_REJECTED);
-+	t38_change_state(session, NULL, datastore->data, T38_REJECTED);
- 	ast_sip_session_resume_reinvite(session);
- 
- 	return 0;
-@@ -322,28 +320,37 @@
- 		int index;
- 
- 		session_media = session->active_media_state->default_session[AST_MEDIA_TYPE_IMAGE];
--		t38_change_state(session, session_media, state, T38_ENABLED);
-+		if (!session_media) {
-+			ast_log(LOG_WARNING, "Received %d response to T.38 re-invite on '%s' but no active session media\n",
-+					status.code, session->channel ? ast_channel_name(session->channel) : "unknown channel");
-+		} else {
-+			t38_change_state(session, session_media, state, T38_ENABLED);
- 
--		/* Stop all the streams in the stored away active state, they'll go back to being active once
--		 * we reinvite back.
--		 */
--		for (index = 0; index < AST_VECTOR_SIZE(&state->media_state->sessions); ++index) {
--			struct ast_sip_session_media *session_media = AST_VECTOR_GET(&state->media_state->sessions, index);
-+			/* Stop all the streams in the stored away active state, they'll go back to being active once
-+			 * we reinvite back.
-+			 */
-+			for (index = 0; index < AST_VECTOR_SIZE(&state->media_state->sessions); ++index) {
-+				struct ast_sip_session_media *session_media = AST_VECTOR_GET(&state->media_state->sessions, index);
- 
--			if (session_media && session_media->handler && session_media->handler->stream_stop) {
--				session_media->handler->stream_stop(session_media);
-+				if (session_media && session_media->handler && session_media->handler->stream_stop) {
-+					session_media->handler->stream_stop(session_media);
-+				}
- 			}
-+
-+			return 0;
- 		}
- 	} else {
- 		session_media = session->pending_media_state->default_session[AST_MEDIA_TYPE_IMAGE];
--		t38_change_state(session, session_media, state, T38_REJECTED);
--
--		/* Abort this attempt at switching to T.38 by resetting the pending state and freeing our stored away active state */
--		ast_sip_session_media_state_free(state->media_state);
--		state->media_state = NULL;
--		ast_sip_session_media_state_reset(session->pending_media_state);
- 	}
- 
-+	/* If no session_media then response contained a declined stream, so disable */
-+	t38_change_state(session, NULL, state, session_media ? T38_REJECTED : T38_DISABLED);
-+
-+	/* Abort this attempt at switching to T.38 by resetting the pending state and freeing our stored away active state */
-+	ast_sip_session_media_state_free(state->media_state);
-+	state->media_state = NULL;
-+	ast_sip_session_media_state_reset(session->pending_media_state);
-+
- 	return 0;
- }
- 
-@@ -426,12 +433,10 @@
- 		/* Negotiation can not take place without a valid max_ifp value. */
- 		if (!parameters->max_ifp) {
- 			if (data->session->t38state == T38_PEER_REINVITE) {
--				session_media = data->session->pending_media_state->default_session[AST_MEDIA_TYPE_IMAGE];
--				t38_change_state(data->session, session_media, state, T38_REJECTED);
-+				t38_change_state(data->session, NULL, state, T38_REJECTED);
- 				ast_sip_session_resume_reinvite(data->session);
- 			} else if (data->session->t38state == T38_ENABLED) {
--				session_media = data->session->active_media_state->default_session[AST_MEDIA_TYPE_IMAGE];
--				t38_change_state(data->session, session_media, state, T38_DISABLED);
-+				t38_change_state(data->session, NULL, state, T38_DISABLED);
- 				ast_sip_session_refresh(data->session, NULL, NULL, NULL,
- 					AST_SIP_SESSION_REFRESH_METHOD_INVITE, 1, state->media_state);
- 				state->media_state = NULL;
-@@ -454,6 +459,11 @@
- 			state->our_parms.version = MIN(state->our_parms.version, state->their_parms.version);
- 			state->our_parms.rate_management = state->their_parms.rate_management;
- 			session_media = data->session->pending_media_state->default_session[AST_MEDIA_TYPE_IMAGE];
-+			if (!session_media) {
-+				ast_log(LOG_ERROR, "Failed to negotiate parameters for reinvite on channel '%s' (No pending session media).\n",
-+					data->session->channel ? ast_channel_name(data->session->channel) : "unknown channel");
-+				break;
-+			}
- 			ast_udptl_set_local_max_ifp(session_media->udptl, state->our_parms.max_ifp);
- 			t38_change_state(data->session, session_media, state, T38_ENABLED);
- 			ast_sip_session_resume_reinvite(data->session);
-@@ -468,8 +478,13 @@
- 			}
- 			state->our_parms = *parameters;
- 			session_media = media_state->default_session[AST_MEDIA_TYPE_IMAGE];
-+			if (!session_media) {
-+				ast_log(LOG_ERROR, "Failed to negotiate parameters on channel '%s' (No default session media).\n",
-+					data->session->channel ? ast_channel_name(data->session->channel) : "unknown channel");
-+				break;
-+			}
- 			ast_udptl_set_local_max_ifp(session_media->udptl, state->our_parms.max_ifp);
--			t38_change_state(data->session, session_media, state, T38_LOCAL_REINVITE);
-+			t38_change_state(data->session, NULL, state, T38_LOCAL_REINVITE);
- 			ast_sip_session_refresh(data->session, NULL, t38_reinvite_sdp_cb, t38_reinvite_response_cb,
- 				AST_SIP_SESSION_REFRESH_METHOD_INVITE, 1, media_state);
- 		}
-@@ -478,12 +493,10 @@
- 	case AST_T38_REFUSED:
- 	case AST_T38_REQUEST_TERMINATE:         /* Shutdown T38 */
- 		if (data->session->t38state == T38_PEER_REINVITE) {
--			session_media = data->session->pending_media_state->default_session[AST_MEDIA_TYPE_IMAGE];
--			t38_change_state(data->session, session_media, state, T38_REJECTED);
-+			t38_change_state(data->session, NULL, state, T38_REJECTED);
- 			ast_sip_session_resume_reinvite(data->session);
- 		} else if (data->session->t38state == T38_ENABLED) {
--			session_media = data->session->active_media_state->default_session[AST_MEDIA_TYPE_IMAGE];
--			t38_change_state(data->session, session_media, state, T38_DISABLED);
-+			t38_change_state(data->session, NULL, state, T38_DISABLED);
- 			ast_sip_session_refresh(data->session, NULL, NULL, NULL, AST_SIP_SESSION_REFRESH_METHOD_INVITE, 1, state->media_state);
- 			state->media_state = NULL;
- 		}
-@@ -493,6 +506,11 @@
- 
- 		if (data->session->t38state == T38_PEER_REINVITE) {
- 			session_media = data->session->pending_media_state->default_session[AST_MEDIA_TYPE_IMAGE];
-+			if (!session_media) {
-+				ast_log(LOG_ERROR, "Failed to request parameters for reinvite on channel '%s' (No pending session media).\n",
-+					data->session->channel ? ast_channel_name(data->session->channel) : "unknown channel");
-+				break;
-+			}
- 			parameters.max_ifp = ast_udptl_get_far_max_ifp(session_media->udptl);
- 			parameters.request_response = AST_T38_REQUEST_NEGOTIATE;
- 			ast_queue_control_data(data->session->channel, AST_CONTROL_T38_PARAMETERS, &parameters, sizeof(parameters));
-@@ -788,7 +806,7 @@
- 
- 	if ((session->t38state == T38_REJECTED) || (session->t38state == T38_DISABLED)) {
- 		ast_debug(3, "Declining; T.38 state is rejected or declined\n");
--		t38_change_state(session, session_media, state, T38_DISABLED);
-+		t38_change_state(session, NULL, state, T38_DISABLED);
- 		return 0;
- 	}
- 
diff --git a/net/asterisk-16.x/patches/170-AST-2019-006-16.diff b/net/asterisk-16.x/patches/170-AST-2019-006-16.diff
deleted file mode 100644
index 1f589b2..0000000
--- a/net/asterisk-16.x/patches/170-AST-2019-006-16.diff
+++ /dev/null
@@ -1,73 +0,0 @@
-From 8cdaa93e658a46e7baf6b606468b5e2c88a0133b Mon Sep 17 00:00:00 2001
-From: Ben Ford <bford@digium.com>
-Date: Mon, 21 Oct 2019 14:55:06 -0500
-Subject: [PATCH] chan_sip.c: Prevent address change on unauthenticated SIP request.
-
-If the name of a peer is known and a SIP request is sent using that
-peer's name, the address of the peer will change even if the request
-fails the authentication challenge. This means that an endpoint can
-be altered and even rendered unusuable, even if it was in a working
-state previously. This can only occur when the nat option is set to the
-default, or auto_force_rport.
-
-This change checks the result of authentication first to ensure it is
-successful before setting the address and the nat option.
-
-ASTERISK-28589 #close
-
-Change-Id: I581c5ed1da60ca89f590bd70872de2b660de02df
----
-
-diff --git a/channels/chan_sip.c b/channels/chan_sip.c
-index 6ac2e61..4d79a47 100644
---- a/channels/chan_sip.c
-+++ b/channels/chan_sip.c
-@@ -19245,18 +19245,6 @@
- 		bogus_peer = NULL;
- 	}
- 
--	/*  build_peer, called through sip_find_peer, is not able to check the
--	 *  sip_pvt->natdetected flag in order to determine if the peer is behind
--	 *  NAT or not when SIP_PAGE3_NAT_AUTO_RPORT or SIP_PAGE3_NAT_AUTO_COMEDIA
--	 *  are set on the peer.  So we check for that here and set the peer's
--	 *  address accordingly.
--	 */
--	set_peer_nat(p, peer);
--
--	if (p->natdetected && ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
--		ast_sockaddr_copy(&peer->addr, &p->recv);
--	}
--
- 	if (!ast_apply_acl(peer->acl, addr, "SIP Peer ACL: ")) {
- 		ast_debug(2, "Found peer '%s' for '%s', but fails host access\n", peer->name, of);
- 		sip_unref_peer(peer, "sip_unref_peer: check_peer_ok: from sip_find_peer call, early return of AUTH_ACL_FAILED");
-@@ -19325,6 +19313,21 @@
- 		ast_string_field_set(p, peermd5secret, NULL);
- 	}
- 	if (!(res = check_auth(p, req, peer->name, p->peersecret, p->peermd5secret, sipmethod, uri2, reliable))) {
-+
-+		/* build_peer, called through sip_find_peer, is not able to check the
-+		 * sip_pvt->natdetected flag in order to determine if the peer is behind
-+		 * NAT or not when SIP_PAGE3_NAT_AUTO_RPORT or SIP_PAGE3_NAT_AUTO_COMEDIA
-+		 * are set on the peer. So we check for that here and set the peer's
-+		 * address accordingly. The address should ONLY be set once we are sure
-+		 * authentication was a success. If, for example, an INVITE was sent that
-+		 * matched the peer name but failed the authentication check, the address
-+		 * would be updated, which is bad.
-+		 */
-+		set_peer_nat(p, peer);
-+		if (p->natdetected && ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
-+			ast_sockaddr_copy(&peer->addr, &p->recv);
-+		}
-+
- 		/* If we have a call limit, set flag */
- 		if (peer->call_limit)
- 			ast_set_flag(&p->flags[0], SIP_CALL_LIMIT);
-@@ -19424,6 +19427,7 @@
- 		}
- 	}
- 	sip_unref_peer(peer, "check_peer_ok: sip_unref_peer: tossing temp ptr to peer from sip_find_peer");
-+
- 	return res;
- }
- 
diff --git a/net/asterisk-16.x/patches/180-AST-2019-007-16.diff b/net/asterisk-16.x/patches/180-AST-2019-007-16.diff
deleted file mode 100644
index 3ae5553..0000000
--- a/net/asterisk-16.x/patches/180-AST-2019-007-16.diff
+++ /dev/null
@@ -1,41 +0,0 @@
-From 7574be5110e049a44b8c8ead52cd1c2a5d442afa Mon Sep 17 00:00:00 2001
-From: George Joseph <gjoseph@digium.com>
-Date: Thu, 24 Oct 2019 11:41:23 -0600
-Subject: [PATCH] manager.c:  Prevent the Originate action from running the Originate app
-
-If an AMI user without the "system" authorization calls the
-Originate AMI command with the Originate application,
-the second Originate could run the "System" command.
-
-Action: Originate
-Channel: Local/1111
-Application: Originate
-Data: Local/2222,app,System,touch /tmp/owned
-
-If the "system" authorization isn't set, we now block the
-Originate app as well as the System, Exec, etc. apps.
-
-ASTERISK-28580
-Reported by: Eliel Sardañons
-
-Change-Id: Ic4c9dedc34c426f03c8c14fce334a71386d8a5fa
----
-
---- /dev/null
-+++ b/doc/UPGRADE-staging/AMI-Originate.txt
-@@ -0,0 +1,5 @@
-+Subject: AMI
-+
-+The AMI Originate action, which optionally takes a dialplan application as
-+an argument, no longer accepts "Originate" as the application due to
-+security concerns.
---- a/main/manager.c
-+++ b/main/manager.c
-@@ -5697,6 +5697,7 @@ static int action_originate(struct manse
- 				                                     EAGI(/bin/rm,-rf /)       */
- 				strcasestr(app, "mixmonitor") ||  /* MixMonitor(blah,,rm -rf)  */
- 				strcasestr(app, "externalivr") || /* ExternalIVR(rm -rf)       */
-+				strcasestr(app, "originate") ||   /* Originate(Local/1234,app,System,rm -rf) */
- 				(strstr(appdata, "SHELL") && (bad_appdata = 1)) ||       /* NoOp(${SHELL(rm -rf /)})  */
- 				(strstr(appdata, "EVAL") && (bad_appdata = 1))           /* NoOp(${EVAL(${some_var_containing_SHELL})}) */
- 				)) {
diff --git a/net/asterisk-16.x/patches/190-AST-2020-001-16.diff b/net/asterisk-16.x/patches/190-AST-2020-001-16.diff
deleted file mode 100644
index dbe58fa..0000000
--- a/net/asterisk-16.x/patches/190-AST-2020-001-16.diff
+++ /dev/null
@@ -1,401 +0,0 @@
-commit 523ed150b16d799bcf223b841abd82f25b8cd6a0
-Author: Kevin Harwell <kharwell@digium.com>
-Date:   Mon Oct 19 17:21:57 2020 -0500
-
-    AST-2020-001 - res_pjsip: Return dialog locked and referenced
-    
-    pjproject returns the dialog locked and with a reference. However,
-    in Asterisk the method that handles this decrements the reference
-    and removes the lock prior to returning. This makes it possible,
-    under some circumstances, for another thread to free said dialog
-    before the thread that created it attempts to use it again. Of
-    course when the thread that created it tries to use a freed dialog
-    a crash can occur.
-    
-    This patch makes it so Asterisk now returns the newly created
-    dialog both locked, and with an added reference. This allows the
-    caller to de-reference, and unlock the dialog when it is safe to
-    do so.
-    
-    In the case of a new SIP Invite the lock, and reference are now
-    held for the entirety of the new invite handling process.
-    Otherwise it's possible for the dialog, or its dependent objects,
-    like the transaction, to disappear. For example if there is a TCP
-    transport error.
-    
-    Change-Id: I5ef645a47829596f402cf383dc02c629c618969e
-
---- a/include/asterisk/res_pjsip.h
-+++ b/include/asterisk/res_pjsip.h
-@@ -1908,6 +1908,11 @@ pjsip_dialog *ast_sip_create_dialog_uac(
- /*!
-  * \brief General purpose method for creating a UAS dialog with an endpoint
-  *
-+ * \deprecated This function is unsafe (due to the returned object not being locked nor
-+ *             having its reference incremented) and should no longer be used. Instead
-+ *             use ast_sip_create_dialog_uas_locked so a properly locked and referenced
-+ *             object is returned.
-+ *
-  * \param endpoint A pointer to the endpoint
-  * \param rdata The request that is starting the dialog
-  * \param[out] status On failure, the reason for failure in creating the dialog
-@@ -1915,6 +1920,44 @@ pjsip_dialog *ast_sip_create_dialog_uac(
- pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata, pj_status_t *status);
- 
- /*!
-+ * \brief General purpose method for creating a UAS dialog with an endpoint
-+ *
-+ * This function creates and returns a locked, and referenced counted pjsip
-+ * dialog object. The caller is thus responsible for freeing the allocated
-+ * memory, decrementing the reference, and releasing the lock when done with
-+ * the returned object.
-+ *
-+ * \note The safest way to unlock the object, and decrement its reference is by
-+ *       calling pjsip_dlg_dec_lock. Alternatively, pjsip_dlg_dec_session can be
-+ *       used to decrement the reference only.
-+ *
-+ * The dialog is returned locked and with a reference in order to ensure that the
-+ * dialog object, and any of its associated objects (e.g. transaction) are not
-+ * untimely destroyed. For instance, that could happen when a transport error
-+ * occurs.
-+ *
-+ * As long as the caller maintains a reference to the dialog there should be no
-+ * worry that it might unknowningly be destroyed. However, once the caller unlocks
-+ * the dialog there is a danger that some of the dialog's internal objects could
-+ * be lost and/or compromised. For example, when the aforementioned transport error
-+ * occurs the dialog's associated transaction gets destroyed (see pjsip_dlg_on_tsx_state
-+ * in sip_dialog.c, and mod_inv_on_tsx_state in sip_inv.c).
-+ *
-+ * In this case and before using the dialog again the caller should re-lock the
-+ * dialog, check to make sure the dialog is still established, and the transaction
-+ * still exists and has not been destroyed.
-+ *
-+ * \param endpoint A pointer to the endpoint
-+ * \param rdata The request that is starting the dialog
-+ * \param[out] status On failure, the reason for failure in creating the dialog
-+ *
-+ * \retval A locked, and reference counted pjsip_dialog object.
-+ * \retval NULL on failure
-+ */
-+pjsip_dialog *ast_sip_create_dialog_uas_locked(const struct ast_sip_endpoint *endpoint,
-+	pjsip_rx_data *rdata, pj_status_t *status);
-+
-+/*!
-  * \brief General purpose method for creating an rdata structure using specific information
-  * \since 13.15.0
-  *
---- a/res/res_pjsip.c
-+++ b/res/res_pjsip.c
-@@ -3645,7 +3645,11 @@ static int uas_use_sips_contact(pjsip_rx
- 	return 0;
- }
- 
--pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata, pj_status_t *status)
-+typedef pj_status_t (*create_dlg_uac)(pjsip_user_agent *ua, pjsip_rx_data *rdata,
-+	const pj_str_t *contact, pjsip_dialog **p_dlg);
-+
-+static pjsip_dialog *create_dialog_uas(const struct ast_sip_endpoint *endpoint,
-+	pjsip_rx_data *rdata, pj_status_t *status, create_dlg_uac create_fun)
- {
- 	pjsip_dialog *dlg;
- 	pj_str_t contact;
-@@ -3680,11 +3684,7 @@ pjsip_dialog *ast_sip_create_dialog_uas(
- 			(type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? ";transport=" : "",
- 			(type != PJSIP_TRANSPORT_UDP && type != PJSIP_TRANSPORT_UDP6) ? pjsip_transport_get_type_name(type) : "");
- 
--#ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK
--	*status = pjsip_dlg_create_uas_and_inc_lock(pjsip_ua_instance(), rdata, &contact, &dlg);
--#else
--	*status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, &contact, &dlg);
--#endif
-+	*status = create_fun(pjsip_ua_instance(), rdata, &contact, &dlg);
- 	if (*status != PJ_SUCCESS) {
- 		char err[PJ_ERR_MSG_SIZE];
- 
-@@ -3697,11 +3697,46 @@ pjsip_dialog *ast_sip_create_dialog_uas(
- 	dlg->sess_count++;
- 	pjsip_dlg_set_transport(dlg, &selector);
- 	dlg->sess_count--;
-+
-+	return dlg;
-+}
-+
-+pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint, pjsip_rx_data *rdata, pj_status_t *status)
-+{
- #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK
--	pjsip_dlg_dec_lock(dlg);
-+	pjsip_dialog *dlg;
-+
-+	dlg = create_dialog_uas(endpoint, rdata, status, pjsip_dlg_create_uas_and_inc_lock);
-+	if (dlg) {
-+		pjsip_dlg_dec_lock(dlg);
-+	}
-+
-+	return dlg;
-+#else
-+	return create_dialog_uas(endpoint, rdata, status, pjsip_dlg_create_uas);
- #endif
-+}
-+
-+pjsip_dialog *ast_sip_create_dialog_uas_locked(const struct ast_sip_endpoint *endpoint,
-+	pjsip_rx_data *rdata, pj_status_t *status)
-+{
-+#ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK
-+	return create_dialog_uas(endpoint, rdata, status, pjsip_dlg_create_uas_and_inc_lock);
-+#else
-+	/*
-+	 * This is put here in order to be compatible with older versions of pjproject.
-+	 * Best we can do in this case is immediately lock after getting the dialog.
-+	 * However, that does leave a "gap" between creating and locking.
-+	 */
-+	pjsip_dialog *dlg;
-+
-+	dlg = create_dialog_uas(endpoint, rdata, status, pjsip_dlg_create_uas);
-+	if (dlg) {
-+		pjsip_dlg_inc_lock(dlg);
-+	}
- 
- 	return dlg;
-+#endif
- }
- 
- int ast_sip_create_rdata_with_contact(pjsip_rx_data *rdata, char *packet, const char *src_name, int src_port,
---- a/res/res_pjsip_pubsub.c
-+++ b/res/res_pjsip_pubsub.c
-@@ -1457,7 +1457,7 @@ static struct sip_subscription_tree *cre
- 	}
- 	sub_tree->role = AST_SIP_NOTIFIER;
- 
--	dlg = ast_sip_create_dialog_uas(endpoint, rdata, dlg_status);
-+	dlg = ast_sip_create_dialog_uas_locked(endpoint, rdata, dlg_status);
- 	if (!dlg) {
- 		if (*dlg_status != PJ_EEXISTS) {
- 			ast_log(LOG_WARNING, "Unable to create dialog for SIP subscription\n");
-@@ -1478,8 +1478,16 @@ static struct sip_subscription_tree *cre
- 	}
- 
- 	pjsip_evsub_create_uas(dlg, &pubsub_cb, rdata, 0, &sub_tree->evsub);
-+
- 	subscription_setup_dialog(sub_tree, dlg);
- 
-+	/*
-+	 * The evsub and subscription setup both add dialog refs, so the dialog ref that
-+	 * was added when the dialog was created (see ast_sip_create_dialog_uas_lock) can
-+	 * now be removed. The lock should no longer be needed so can be removed too.
-+	 */
-+	pjsip_dlg_dec_lock(dlg);
-+
- #ifdef HAVE_PJSIP_EVSUB_GRP_LOCK
- 	pjsip_evsub_add_ref(sub_tree->evsub);
- #endif
---- a/res/res_pjsip_session.c
-+++ b/res/res_pjsip_session.c
-@@ -2942,6 +2942,75 @@ static enum sip_get_destination_result g
- 	return SIP_GET_DEST_EXTEN_NOT_FOUND;
- }
- 
-+/*
-+ * /internal
-+ * /brief Process initial answer for an incoming invite
-+ *
-+ * This function should only be called during the setup, and handling of a
-+ * new incoming invite. Most, if not all of the time, this will be called
-+ * when an error occurs and we need to respond as such.
-+ *
-+ * When a SIP session termination code is given for the answer it's assumed
-+ * this call then will be the final bit of processing before ending session
-+ * setup. As such, we've been holding a lock, and a reference on the invite
-+ * session's dialog. So before returning this function removes that reference,
-+ * and unlocks the dialog.
-+ *
-+ * \param inv_session The session on which to answer
-+ * \param rdata The original request
-+ * \param answer_code The answer's numeric code
-+ * \param terminate_code The termination code if the answer fails
-+ * \param notify Whether or not to call on_state_changed
-+ *
-+ * \retval 0 if invite successfully answered, -1 if an error occurred
-+ */
-+static int new_invite_initial_answer(pjsip_inv_session *inv_session, pjsip_rx_data *rdata,
-+	int answer_code, int terminate_code, pj_bool_t notify)
-+{
-+	pjsip_tx_data *tdata = NULL;
-+	int res = 0;
-+
-+	if (inv_session->state != PJSIP_INV_STATE_DISCONNECTED) {
-+		if (pjsip_inv_initial_answer(
-+				inv_session, rdata, answer_code, NULL, NULL, &tdata) != PJ_SUCCESS) {
-+
-+			pjsip_inv_terminate(inv_session, terminate_code ? terminate_code : answer_code, notify);
-+			res = -1;
-+		} else {
-+			pjsip_inv_send_msg(inv_session, tdata);
-+		}
-+	}
-+
-+	if (answer_code >= 300) {
-+		/*
-+		 * A session is ending. The dialog has a reference that needs to be
-+		 * removed and holds a lock that needs to be unlocked before returning.
-+		 */
-+		pjsip_dlg_dec_lock(inv_session->dlg);
-+	}
-+
-+	return res;
-+}
-+
-+/*
-+ * /internal
-+ * /brief Create and initialize a pjsip invite session
-+
-+ * pjsip_inv_session adds, and maintains a reference to the dialog upon a successful
-+ * invite session creation until the session is destroyed. However, we'll wait to
-+ * remove the reference that was added for the dialog when it gets created since we're
-+ * not ready to unlock the dialog in this function.
-+ *
-+ * So, if this function successfully returns that means it returns with its newly
-+ * created, and associated dialog locked and with two references (i.e. dialog's
-+ * reference count should be 2).
-+ *
-+ * \param endpoint A pointer to the endpoint
-+ * \param rdata The request that is starting the dialog
-+ *
-+ * \retval A pjsip invite session object
-+ * \retval NULL on error
-+ */
- static pjsip_inv_session *pre_session_setup(pjsip_rx_data *rdata, const struct ast_sip_endpoint *endpoint)
- {
- 	pjsip_tx_data *tdata;
-@@ -2960,15 +3029,28 @@ static pjsip_inv_session *pre_session_se
- 		}
- 		return NULL;
- 	}
--	dlg = ast_sip_create_dialog_uas(endpoint, rdata, &dlg_status);
-+
-+	dlg = ast_sip_create_dialog_uas_locked(endpoint, rdata, &dlg_status);
- 	if (!dlg) {
- 		if (dlg_status != PJ_EEXISTS) {
- 			pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 500, NULL, NULL, NULL);
- 		}
- 		return NULL;
- 	}
-+
-+	/*
-+	 * The returned dialog holds a lock and has a reference added. Any paths where the
-+	 * dialog invite session is not returned must unlock the dialog and remove its reference.
-+	 */
-+
- 	if (pjsip_inv_create_uas(dlg, rdata, NULL, options, &inv_session) != PJ_SUCCESS) {
- 		pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 500, NULL, NULL, NULL);
-+		/*
-+		 * The acquired dialog holds a lock, and a reference. Since the dialog is not
-+		 * going to be returned here it must first be unlocked and de-referenced. This
-+		 * must be done prior to calling dialog termination.
-+		 */
-+		pjsip_dlg_dec_lock(dlg);
- 		pjsip_dlg_terminate(dlg);
- 		return NULL;
- 	}
-@@ -2977,12 +3059,13 @@ static pjsip_inv_session *pre_session_se
- 	inv_session->sdp_neg_flags = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE;
- #endif
- 	if (pjsip_dlg_add_usage(dlg, &session_module, NULL) != PJ_SUCCESS) {
--		if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) != PJ_SUCCESS) {
--			pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
--		}
--		pjsip_inv_send_msg(inv_session, tdata);
-+		/* Dialog's lock and a reference are removed in new_invite_initial_answer */
-+		new_invite_initial_answer(inv_session, rdata, 500, 500, PJ_FALSE);
-+		/* Remove 2nd reference added at inv_session creation */
-+		pjsip_dlg_dec_session(inv_session->dlg, &session_module);
- 		return NULL;
- 	}
-+
- 	return inv_session;
- }
- 
-@@ -3121,7 +3204,6 @@ static void handle_new_invite_request(pj
- {
- 	RAII_VAR(struct ast_sip_endpoint *, endpoint,
- 			ast_pjsip_rdata_get_endpoint(rdata), ao2_cleanup);
--	pjsip_tx_data *tdata = NULL;
- 	pjsip_inv_session *inv_session = NULL;
- 	struct ast_sip_session *session;
- 	struct new_invite invite;
-@@ -3134,27 +3216,48 @@ static void handle_new_invite_request(pj
- 		return;
- 	}
- 
-+	/*
-+	 * Upon a successful pre_session_setup the associated dialog is returned locked
-+	 * and with an added reference. Well actually two references. One added when the
-+	 * dialog itself was created, and another added when the pjsip invite session was
-+	 * created and the dialog was added to it.
-+	 *
-+	 * In order to ensure the dialog's, and any of its internal attributes, lifetimes
-+	 * we'll hold the lock and maintain the reference throughout the entire new invite
-+	 * handling process. See ast_sip_create_dialog_uas_locked for more details but,
-+	 * basically we do this to make sure a transport failure does not destroy the dialog
-+	 * and/or transaction out from underneath us between pjsip calls. Alternatively, we
-+	 * could probably release the lock if we needed to, but then we'd have to re-lock and
-+	 * check the dialog and transaction prior to every pjsip call.
-+	 *
-+	 * That means any off nominal/failure paths in this function must remove the associated
-+	 * dialog reference added at dialog creation, and remove the lock. As well the
-+	 * referenced pjsip invite session must be "cleaned up", which should also then
-+	 * remove its reference to the dialog at that time.
-+	 *
-+	 * Nominally we'll unlock the dialog, and release the reference when all new invite
-+	 * process handling has successfully completed.
-+	 */
-+
- #ifdef HAVE_PJSIP_INV_SESSION_REF
- 	if (pjsip_inv_add_ref(inv_session) != PJ_SUCCESS) {
- 		ast_log(LOG_ERROR, "Can't increase the session reference counter\n");
--		if (inv_session->state != PJSIP_INV_STATE_DISCONNECTED) {
--			if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) == PJ_SUCCESS) {
--				pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
--			} else {
--				pjsip_inv_send_msg(inv_session, tdata);
--			}
-+		/* Dialog's lock and a reference are removed in new_invite_initial_answer */
-+		if (!new_invite_initial_answer(inv_session, rdata, 500, 500, PJ_FALSE)) {
-+			/* Terminate the session if it wasn't done in the answer */
-+			pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
- 		}
- 		return;
- 	}
- #endif
--
- 	session = ast_sip_session_alloc(endpoint, NULL, inv_session, rdata);
- 	if (!session) {
--		if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) == PJ_SUCCESS) {
-+		/* Dialog's lock and reference are removed in new_invite_initial_answer */
-+		if (!new_invite_initial_answer(inv_session, rdata, 500, 500, PJ_FALSE)) {
-+			/* Terminate the session if it wasn't done in the answer */
- 			pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
--		} else {
--			pjsip_inv_send_msg(inv_session, tdata);
- 		}
-+
- #ifdef HAVE_PJSIP_INV_SESSION_REF
- 		pjsip_inv_dec_ref(inv_session);
- #endif
-@@ -3172,6 +3275,17 @@ static void handle_new_invite_request(pj
- 	invite.rdata = rdata;
- 	new_invite(&invite);
- 
-+	/*
-+	 * The dialog lock and reference added at dialog creation time must be
-+	 * maintained throughout the new invite process. Since we're pretty much
-+	 * done at this point with things it's safe to go ahead and remove the lock
-+	 * and the reference here. See ast_sip_create_dialog_uas_locked for more info.
-+	 *
-+	 * Note, any future functionality added that does work using the dialog must
-+	 * be done before this.
-+	 */
-+	pjsip_dlg_dec_lock(inv_session->dlg);
-+
- 	ao2_ref(session, -1);
- }
- 
diff --git a/net/asterisk-16.x/patches/200-AST-2020-002-16.diff b/net/asterisk-16.x/patches/200-AST-2020-002-16.diff
deleted file mode 100644
index 010ea34..0000000
--- a/net/asterisk-16.x/patches/200-AST-2020-002-16.diff
+++ /dev/null
@@ -1,102 +0,0 @@
-From f83efa12f7411dd100f49933bf71297c8ed9f765 Mon Sep 17 00:00:00 2001
-From: Ben Ford <bford@digium.com>
-Date: Mon, 02 Nov 2020 10:29:31 -0600
-Subject: [PATCH] AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit.
-
-If Asterisk sends out an INVITE and receives a challenge with a
-different nonce value each time, it will continuously send out INVITEs,
-even if the call is hung up. The endpoint must be configured for
-outbound authentication for this to occur. A limit has been set on
-outbound INVITEs so that, once reached, Asterisk will stop sending
-INVITEs and the transaction will terminate.
-
-ASTERISK-29013
-
-Change-Id: I2d001ca745b00ca8aa12030f2240cd72363b46f7
----
-
---- a/include/asterisk/res_pjsip.h
-+++ b/include/asterisk/res_pjsip.h
-@@ -63,6 +63,9 @@ struct pjsip_tpselector;
- /*! \brief Maximum number of ciphers supported for a TLS transport */
- #define SIP_TLS_MAX_CIPHERS 64
- 
-+/*! Maximum number of challenges before assuming that we are in a loop */
-+#define MAX_RX_CHALLENGES	10
-+
- /*!
-  * \brief Structure for SIP transport information
-  */
---- a/include/asterisk/res_pjsip_session.h
-+++ b/include/asterisk/res_pjsip_session.h
-@@ -215,8 +215,10 @@ struct ast_sip_session {
- 	enum ast_sip_dtmf_mode dtmf;
- 	/*! Initial incoming INVITE Request-URI.  NULL otherwise. */
- 	pjsip_uri *request_uri;
--	/* Media statistics for negotiated RTP streams */
-+	/*! Media statistics for negotiated RTP streams */
- 	AST_VECTOR(, struct ast_rtp_instance_stats *) media_stats;
-+	/*! Number of challenges received during outgoing requests to determine if we are in a loop */
-+	unsigned int authentication_challenge_count:4;
- };
- 
- typedef int (*ast_sip_session_request_creation_cb)(struct ast_sip_session *session, pjsip_tx_data *tdata);
---- a/res/res_pjsip.c
-+++ b/res/res_pjsip.c
-@@ -4027,8 +4027,6 @@ static pj_bool_t does_method_match(const
- 	return pj_stristr(&method, message_method) ? PJ_TRUE : PJ_FALSE;
- }
- 
--/*! Maximum number of challenges before assuming that we are in a loop */
--#define MAX_RX_CHALLENGES	10
- #define TIMER_INACTIVE		0
- #define TIMEOUT_TIMER2		5
- 
---- a/res/res_pjsip_session.c
-+++ b/res/res_pjsip_session.c
-@@ -2052,7 +2052,6 @@ static pjsip_module session_reinvite_mod
- 	.on_rx_request = session_reinvite_on_rx_request,
- };
- 
--
- void ast_sip_session_send_request_with_cb(struct ast_sip_session *session, pjsip_tx_data *tdata,
- 		ast_sip_session_response_cb on_response)
- {
-@@ -2301,6 +2300,9 @@ struct ast_sip_session *ast_sip_session_
- 		return NULL;
- 	}
- 
-+	/* Track the number of challenges received on outbound requests */
-+	session->authentication_challenge_count = 0;
-+
- 	/* Fire seesion begin handlers */
- 	handle_session_begin(session);
- 
-@@ -2470,6 +2472,11 @@ static pj_bool_t outbound_invite_auth(pj
- 
- 	session = inv->mod_data[session_module.id];
- 
-+	if (++session->authentication_challenge_count > MAX_RX_CHALLENGES) {
-+		ast_debug(1, "Initial INVITE reached maximum number of auth attempts.\n");
-+		return PJ_FALSE;
-+	}
-+
- 	if (ast_sip_create_request_with_auth(&session->endpoint->outbound_auths, rdata,
- 		tsx->last_tx, &tdata)) {
- 		return PJ_FALSE;
-@@ -3846,6 +3853,7 @@ static void session_inv_on_tsx_state_cha
- 						ast_debug(1, "reINVITE received final response code %d\n",
- 							tsx->status_code);
- 						if ((tsx->status_code == 401 || tsx->status_code == 407)
-+							&& ++session->authentication_challenge_count < MAX_RX_CHALLENGES
- 							&& !ast_sip_create_request_with_auth(
- 								&session->endpoint->outbound_auths,
- 								e->body.tsx_state.src.rdata, tsx->last_tx, &tdata)) {
-@@ -3920,6 +3928,7 @@ static void session_inv_on_tsx_state_cha
- 						(int) pj_strlen(&tsx->method.name), pj_strbuf(&tsx->method.name),
- 						tsx->status_code);
- 					if ((tsx->status_code == 401 || tsx->status_code == 407)
-+						&& ++session->authentication_challenge_count < MAX_RX_CHALLENGES
- 						&& !ast_sip_create_request_with_auth(
- 							&session->endpoint->outbound_auths,
- 							e->body.tsx_state.src.rdata, tsx->last_tx, &tdata)) {
diff --git a/net/asterisk-16.x/patches/210-AST-2021-001-16.diff b/net/asterisk-16.x/patches/210-AST-2021-001-16.diff
deleted file mode 100644
index 9bb13d3..0000000
--- a/net/asterisk-16.x/patches/210-AST-2021-001-16.diff
+++ /dev/null
@@ -1,87 +0,0 @@
-From 757b7f8d7cfee4f541e8d7586e2408556a74201d Mon Sep 17 00:00:00 2001
-From: Ivan Poddubnyi <ivan.poddubny@gmail.com>
-Date: Mon, 28 Dec 2020 13:43:23 +0100
-Subject: [PATCH] res_pjsip_diversion: Fix adding more than one histinfo to
- Supported
-
-New responses sent within a PJSIP sessions are based on those that were
-sent before. Therefore, adding/modifying a header once causes it to be
-sent on all responses that follow.
-
-Sending 181 Call Is Being Forwarded many times first adds "histinfo"
-duplicated more and more, and eventually overflows past the array
-boundary.
-
-This commit adds a check preventing adding "histinfo" more than once,
-and skipping it if there is no more space in the header.
-
-Similar overflow situations can also occur in res_pjsip_path and
-res_pjsip_outbound_registration so those were also modified to
-check the bounds and suppress duplicate Supported values.
-
-ASTERISK-29227
-Reported by: Ivan Poddubny
-
-Change-Id: Id43704a1f1a0293e35cc7f844026f0b04f2ac322
----
- res/res_pjsip_diversion.c             | 14 ++++++++++++++
- res/res_pjsip_outbound_registration.c | 12 ++++++++++++
- res/res_pjsip_path.c                  | 12 ++++++++++++
- 3 files changed, 38 insertions(+)
-
---- a/res/res_pjsip_outbound_registration.c
-+++ b/res/res_pjsip_outbound_registration.c
-@@ -580,6 +580,7 @@ static int handle_client_registration(vo
- 
- 	if (client_state->support_path) {
- 		pjsip_supported_hdr *hdr;
-+		int i;
- 
- 		hdr = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_SUPPORTED, NULL);
- 		if (!hdr) {
-@@ -593,6 +594,17 @@ static int handle_client_registration(vo
- 			pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *)hdr);
- 		}
- 
-+		/* Don't add the value if it's already there */
-+		for (i = 0; i < hdr->count; ++i) {
-+			if (pj_stricmp(&hdr->values[i], &PATH_NAME) == 0) {
-+				return 1;
-+			}
-+		}
-+
-+		if (hdr->count >= PJSIP_GENERIC_ARRAY_MAX_COUNT) {
-+			return 0;
-+		}
-+
- 		/* add on to the existing Supported header */
- 		pj_strassign(&hdr->values[hdr->count++], &PATH_NAME);
- 	}
---- a/res/res_pjsip_path.c
-+++ b/res/res_pjsip_path.c
-@@ -122,6 +122,7 @@ static int path_get_string(pj_pool_t *po
- static int add_supported(pjsip_tx_data *tdata)
- {
- 	pjsip_supported_hdr *hdr;
-+	int i;
- 
- 	hdr = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_SUPPORTED, NULL);
- 	if (!hdr) {
-@@ -134,6 +135,17 @@ static int add_supported(pjsip_tx_data *
- 		pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *)hdr);
- 	}
- 
-+	/* Don't add the value if it's already there */
-+	for (i = 0; i < hdr->count; ++i) {
-+		if (pj_stricmp(&hdr->values[i], &PATH_SUPPORTED_NAME) == 0) {
-+			return 0;
-+		}
-+	}
-+
-+	if (hdr->count >= PJSIP_GENERIC_ARRAY_MAX_COUNT) {
-+		return -1;
-+	}
-+
- 	/* add on to the existing Supported header */
- 	pj_strassign(&hdr->values[hdr->count++], &PATH_SUPPORTED_NAME);
-