From abc7a1714211cdea4ad2afd158746a5b14a8c5a0 Mon Sep 17 00:00:00 2001
From: Sebastian Kemper <sebastian_ml@gmx.net>
Date: Wed, 3 Nov 2021 21:43:57 +0100
Subject: [PATCH] asterisk-16.x: bump to 16.22.0

- add new modules
- add res_timing_timerfd to base package (see commit e538fc3)
- update some module dependencies
- refresh patches
- remove upstreamed patches

Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
---
 net/asterisk-16.x/Makefile                    |  44 +-
 ...semaphores-on-uclibc-otherwise-allow.patch |   2 +-
 ...tection-of-re-entrant-resolver-funct.patch |   2 +-
 .../patches/056-fix-check_expr2-build.patch   |  38 --
 .../patches/100-build-reproducibly.patch      |   8 +-
 net/asterisk-16.x/patches/120-loader.patch    | 247 -----------
 net/asterisk-16.x/patches/130-eventfd.patch   |   2 +-
 .../patches/140-AST-2019-002-16.diff          |  40 --
 .../patches/150-AST-2019-003-16.diff          |  39 --
 .../patches/160-AST-2019-004-16.patch         | 171 --------
 .../patches/170-AST-2019-006-16.diff          |  73 ----
 .../patches/180-AST-2019-007-16.diff          |  41 --
 .../patches/190-AST-2020-001-16.diff          | 401 ------------------
 .../patches/200-AST-2020-002-16.diff          | 102 -----
 .../patches/210-AST-2021-001-16.diff          |  87 ----
 15 files changed, 40 insertions(+), 1257 deletions(-)
 delete mode 100644 net/asterisk-16.x/patches/056-fix-check_expr2-build.patch
 delete mode 100644 net/asterisk-16.x/patches/120-loader.patch
 delete mode 100644 net/asterisk-16.x/patches/140-AST-2019-002-16.diff
 delete mode 100644 net/asterisk-16.x/patches/150-AST-2019-003-16.diff
 delete mode 100644 net/asterisk-16.x/patches/160-AST-2019-004-16.patch
 delete mode 100644 net/asterisk-16.x/patches/170-AST-2019-006-16.diff
 delete mode 100644 net/asterisk-16.x/patches/180-AST-2019-007-16.diff
 delete mode 100644 net/asterisk-16.x/patches/190-AST-2020-001-16.diff
 delete mode 100644 net/asterisk-16.x/patches/200-AST-2020-002-16.diff
 delete mode 100644 net/asterisk-16.x/patches/210-AST-2021-001-16.diff

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);
- 
-- 
2.30.2