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asterisk: bump to 22.7.0

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remove deprecated modules from 20-21
21-22 is mostly bugfixes

See https://docs.asterisk.org/Asterisk_21_Documentation/Upgrading/
and https://docs.asterisk.org/Asterisk_22_Documentation/Upgrading/
and https://docs.asterisk.org/Development/Asterisk-Module-Deprecations/#removed-modules

- remove deprecated users.conf (otherwise asterisk logs:
  WARNING[24699] pbx_config.c: users.conf is deprecated and will be removed in
  a future version of Asterisk )
- add cdr.conf (otherwise asterisk logs:
  ERROR[24658] config_options.c: Unable to load config file 'cdr.conf' )
- add missing res-websocket-client (necessary for app_stasis, res_ari)
- app_macro -> app_stack

refresh patches

Signed-off-by: Paul Donald <newtwen+github@gmail.com>
This commit is contained in:
Paul Donald
2025-12-03 23:43:36 +01:00
committed by Jiri Slachta
parent 37ec321f5f
commit 73fd665261
5 changed files with 16 additions and 25 deletions
Download Patch File Download Diff File Expand all files Collapse all files

33
net/asterisk/Makefile
View File

@@ -8,13 +8,13 @@
include $(TOPDIR)/rules.mk
PKG_NAME:=asterisk
PKG_VERSION:=20.14.1
PKG_VERSION:=22.7.0
PKG_RELEASE:=1
PKG_CPE_ID:=cpe:/a:digium:asterisk
PKG_SOURCE:=asterisk-$(PKG_VERSION).tar.gz
PKG_SOURCE_URL:=https://downloads.asterisk.org/pub/telephony/asterisk/releases
PKG_HASH:=fa0f953740eed079d5aaadf88f7f7131a053c61e4bc961faed0f30ba77f52ac9
PKG_HASH:=5b0e5d276aeb014bf8a08a94d1055a9e97b9dce3375846eea70da5221bf9efe7
PKG_BUILD_DEPENDS:=libxml2/host
@@ -125,19 +125,15 @@ MODULES_AVAILABLE:= \
cel-custom \
cel-manager \
cel-sqlite3-custom \
chan-alsa \
chan-audiosocket \
chan-bridge-media \
chan-console \
chan-dahdi \
chan-iax2 \
chan-mgcp \
chan-mobile \
chan-motif \
chan-ooh323 \
chan-rtp \
chan-sip \
chan-skinny \
chan-unistim \
codec-a-mu \
codec-adpcm \
@@ -280,7 +276,6 @@ MODULES_AVAILABLE:= \
res-limit \
res-manager-devicestate \
res-manager-presencestate \
res-monitor \
res-musiconhold \
res-mutestream \
res-mwi-devstate \
@@ -294,7 +289,6 @@ MODULES_AVAILABLE:= \
res-pjsip-rfc3329 \
res-pjsip-stir-shaken \
res-pjproject \
res-pktccops \
res-prometheus \
res-realtime \
res-remb-modifier \
@@ -321,7 +315,9 @@ MODULES_AVAILABLE:= \
res-stun-monitor \
res-timing-dahdi \
res-timing-pthread \
res-timing-timerfd \
res-tonedetect \
res-websocket-client \
res-xmpp
UTILS_AVAILABLE:= \
@@ -482,6 +478,7 @@ endef
define Package/$(PKG_NAME)/conffiles
/etc/asterisk/asterisk.conf
/etc/asterisk/acl.conf
/etc/asterisk/cdr.conf
/etc/asterisk/cel.conf
/etc/asterisk/ccss.conf
/etc/asterisk/cli.conf
@@ -500,20 +497,19 @@ define Package/$(PKG_NAME)/conffiles
/etc/asterisk/res_config_sqlite3.conf
/etc/asterisk/stasis.conf
/etc/asterisk/udptl.conf
/etc/asterisk/users.conf
/etc/config/asterisk
/etc/init.d/asterisk
endef
AST_CFG_FILES:= \
asterisk.conf acl.conf cel.conf ccss.conf cli.conf \
asterisk.conf acl.conf cdr.conf cel.conf ccss.conf cli.conf \
cli_permissions.conf codecs.conf dnsmgr.conf dsp.conf extconfig.conf \
extensions.conf features.conf http.conf indications.conf \
logger.conf manager.conf modules.conf stasis.conf udptl.conf \
users.conf res_config_sqlite3.conf
res_config_sqlite3.conf
AST_EMB_MODULES:=\
app_dial app_echo app_macro app_playback \
app_dial app_echo app_stack app_playback \
func_callerid func_logic func_strings func_timeout \
pbx_config res_crypto res_timing_timerfd
@@ -567,13 +563,12 @@ else
--without-tonezone
endif
# Pass CPPFLAGS in the CFLAGS as otherwise the build system will
# Pass CPPFLAGS in the CFLAGS as otherwise the build system will
# ignore them.
TARGET_CFLAGS+=$(TARGET_CPPFLAGS)
CONFIGURE_ARGS+= \
--disable-xmldoc \
$(if $(CONFIG_PACKAGE_$(PKG_NAME)-chan-alsa),--with-asound="$(STAGING_DIR)/usr",--without-asound) \
--without-execinfo \
$(if $(CONFIG_PACKAGE_$(PKG_NAME)-chan-mobile),--with-bluetooth="$(STAGING_DIR)/usr",--without-bluetooth) \
--with-cap="$(STAGING_DIR)/usr" \
@@ -862,23 +857,19 @@ $(eval $(call BuildAsteriskModule,bridge-simple,Simple two channel bridging modu
$(eval $(call BuildAsteriskModule,bridge-softmix,Multi-party software based channel mixing,Multi-party software based channel mixing.,,,bridge_softmix,,))
$(eval $(call BuildAsteriskModule,cdr,Provides CDR,Call Detail Records.,,cdr.conf cdr_custom.conf cdr_manager.conf,app_cdr app_forkcdr cdr_custom cdr_manager func_cdr,,))
$(eval $(call BuildAsteriskModule,cdr-csv,Provides CDR CSV,Comma Separated Values CDR backend.,,,cdr_csv,,))
$(eval $(call BuildAsteriskModule,cdr-sqlite3,Provides CDR SQLITE3,SQLite3 CDR backend.,libsqlite3,,cdr_sqlite3_custom,,))
$(eval $(call BuildAsteriskModule,cdr-sqlite3,Provides CDR SQLITE3,SQLite3 CDR backend.,libsqlite3,cdr_sqlite3_custom.conf,cdr_sqlite3_custom,,))
$(eval $(call BuildAsteriskModule,cel-custom,Customizable CSV CEL backend,Customizable Comma Separated Values CEL backend.,,cel_custom.conf,cel_custom,,))
$(eval $(call BuildAsteriskModule,cel-manager,AMI CEL backend,Asterisk Manager Interface CEL backend.,,,cel_manager,,))
$(eval $(call BuildAsteriskModule,cel-sqlite3-custom,SQLite3 custom CEL,SQLite3 custom CEL module.,,cel_sqlite3_custom.conf,cel_sqlite3_custom,,))
$(eval $(call BuildAsteriskModule,chan-alsa,ALSA channel,ALSA console channel driver.,+alsa-lib,alsa.conf,chan_alsa,,))
$(eval $(call BuildAsteriskModule,chan-audiosocket,Audiosocket channel,Audiosocket channel.,+$(PKG_NAME)-res-audiosocket,,chan_audiosocket,,))
$(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.,,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-rtp,RTP media channel,RTP media channel.,+asterisk-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,,))
$(eval $(call BuildAsteriskModule,codec-a-mu,Alaw to ulaw translation,Alaw and ulaw direct coder/decoder.,,,codec_a_mu,,))
$(eval $(call BuildAsteriskModule,codec-adpcm,ADPCM text,Adaptive Differential PCM coder/decoder.,,,codec_adpcm,,))
@@ -1021,7 +1012,6 @@ $(eval $(call BuildAsteriskModule,res-http-websocket,HTTP websocket,HTTP WebSock
$(eval $(call BuildAsteriskModule,res-limit,Resource limits,Resource limits.,,,res_limit,,))
$(eval $(call BuildAsteriskModule,res-manager-devicestate,Device state topic forwarder,Manager device state topic forwarder.,,,res_manager_devicestate,,))
$(eval $(call BuildAsteriskModule,res-manager-presencestate,Presence state topic forwarder,Manager presence state topic forwarder.,,,res_manager_presencestate,,))
$(eval $(call BuildAsteriskModule,res-monitor,PBX channel monitoring,Call monitoring resource.,,,res_monitor,,))
$(eval $(call BuildAsteriskModule,res-musiconhold,MOH,Music On Hold resource.,,musiconhold.conf,res_musiconhold,,))
$(eval $(call BuildAsteriskModule,res-mutestream,Mute audio stream resources,Mute audio stream resources.,,,res_mutestream,,))
$(eval $(call BuildAsteriskModule,res-mwi-devstate,MWI device state subs,This module allows presence subscriptions to voicemail boxes. This\nallows common BLF keys to act as voicemail waiting indicators.,,,res_mwi_devstate,,))
@@ -1035,7 +1025,6 @@ $(eval $(call BuildAsteriskModule,res-pjsip-phoneprov,PJSIP Phone Provisioning,P
$(eval $(call BuildAsteriskModule,res-pjsip-rfc3329,PJSIP RFC3329,PJSIP RFC3329 Support (partial).,+asterisk-pjsip +asterisk-res-pjproject,,res_pjsip_rfc3329,,))
$(eval $(call BuildAsteriskModule,res-pjsip-stir-shaken,PJSIP STIR/SHAKEN resource module,PJSIP STIR/SHAKEN resource module.,+$(PKG_NAME)-pjsip +$(PKG_NAME)-res-stir-shaken,,res_pjsip_stir_shaken,,))
$(eval $(call BuildAsteriskModule,res-pjproject,Bridge PJPROJECT to Asterisk logging,PJProject log and utility support.,+asterisk-res-sorcery +libpj +libpjlib-util +libpjmedia +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsip +libpjsua +libpjsua2,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-prometheus,Prometheus resource module,Prometheus resource module.,+libpjsip +libpjmedia +libpjnath +libpjsip-simple +libpjsip-ua +libpjsua +libpjsua2,prometheus.conf,res_prometheus,,))
$(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,,))
@@ -1062,7 +1051,9 @@ $(eval $(call BuildAsteriskModule,res-stir-shaken,STIR/SHAKEN resource module,ST
$(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-websocket-client,WebSocket Client,WebSocket Client module.,,websocket_client.conf,res_websocket_client,,))
$(eval $(call BuildAsteriskModule,res-xmpp,XMPP client and component module,Asterisk XMPP interface.,+libiksemel +libopenssl,xmpp.conf,res_xmpp,,))
################################

2
net/asterisk/patches/053-musl-mutex-init.patch
View File

@@ -1,6 +1,6 @@
--- a/include/asterisk/lock.h
+++ b/include/asterisk/lock.h
@@ -70,7 +70,7 @@ extern "C" {
@@ -73,7 +73,7 @@ extern "C" {
#define AST_PTHREADT_NULL (pthread_t) -1
#define AST_PTHREADT_STOP (pthread_t) -2

2
net/asterisk/patches/130-eventfd.patch
View File

@@ -1,6 +1,6 @@
--- a/configure.ac
+++ b/configure.ac
@@ -1293,7 +1293,7 @@ if test "${ac_cv_have_variable_fdset}x"
@@ -1291,7 +1291,7 @@ if test "${ac_cv_have_variable_fdset}x"
fi
AC_MSG_CHECKING([if we have usable eventfd support])

2
net/asterisk/patches/140-use-default-lua.patch
View File

@@ -1,6 +1,6 @@
--- a/configure.ac
+++ b/configure.ac
@@ -2645,7 +2645,7 @@ if test -z "$__opus_include" -o x"$__opu
@@ -2641,7 +2641,7 @@ if test -z "$__opus_include" -o x"$__opu
fi
AST_EXT_LIB_CHECK([OPUSFILE], [opusfile], [op_open_callbacks], [opus/opusfile.h], [], [$__opus_include])

2
net/asterisk/patches/180-app_queue_time_t.patch
View File

@@ -1,6 +1,6 @@
--- a/apps/app_queue.c
+++ b/apps/app_queue.c
@@ -4800,8 +4800,12 @@ static int is_longest_waiting_caller(str
@@ -4803,8 +4803,12 @@ static int is_longest_waiting_caller(str
* will be unused until the first caller is picked up.
*/
if (ch->start < caller->start && !ch->pending) {