head 1.9; access; symbols pkgsrc-2026Q2:1.8.0.2 pkgsrc-2026Q2-base:1.8 pkgsrc-2026Q1:1.5.0.2 pkgsrc-2026Q1-base:1.5 pkgsrc-2025Q4:1.2.0.2 pkgsrc-2025Q4-base:1.2; locks; strict; comment @# @; 1.9 date 2026.06.22.02.59.47; author jnemeth; state Exp; branches; next 1.8; commitid g55PIxsOKA6xZJKG; 1.8 date 2026.05.14.16.40.32; author ryoon; state Exp; branches; next 1.7; commitid tKipFjQKzke3NNFG; 1.7 date 2026.04.13.03.47.11; author jnemeth; state Exp; branches; next 1.6; commitid 7bTLGN23nxCOuKBG; 1.6 date 2026.04.10.08.41.35; author wiz; state Exp; branches; next 1.5; commitid 4LhvV4JecdcpeoBG; 1.5 date 2026.02.16.04.11.13; author jnemeth; state Exp; branches; next 1.4; commitid w1vHMO4LkZx8qyuG; 1.4 date 2026.02.06.10.04.21; author wiz; state Exp; branches; next 1.3; commitid MwQEYCXeWSFvIitG; 1.3 date 2026.01.07.08.46.04; author wiz; state Exp; branches; next 1.2; commitid 1wQ3ICD8eebefrpG; 1.2 date 2025.12.01.04.26.02; author jnemeth; state Exp; branches; next 1.1; commitid 7MHnImsud9wNZEkG; 1.1 date 2025.10.27.06.58.36; author jnemeth; state Exp; branches; next ; commitid cQXB8JNvDKKmYagG; desc @@ 1.9 log @Update to Asterisk 23.4.0: ## Change Log for Release asterisk-23.4.0 ### Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.4.0.html) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.3.0...23.4.0) ### Summary: - Commits: 53 - Commit Authors: 24 - Issues Resolved: 43 - Security Advisories Resolved: 0 ### User Notes: - #### res ari: Add attachable states to Channels and Bridges Bridge variables now can be set and retrieved via the following paths: `/bridges/{bridgeId}/variable` `/bridges/{bridgeId}/variables` Both Bridge and Channel variables can now be set with an optional 'report_events' boolean flag that will cause those variables to be included on all events on that object. The 'report_events' flag will default to False if not set to maintain backwards capability. To allow this, variables can now be either name value pairs (the current format): `: ''` - or - `: {value: '', report_events: [true|false]}` - #### ARI: Added paths to get and set multiple channel variables. Added new ARI paths for getting and setting multiple channel variables at a time. For GET, this takes in a single string of comma-separated variable names, while POST takes in a dictionary of key value pairs. The behavior is the same as passing in variables when originating a channel. - #### res_rtp_asterisk: Add option to control stun host resolution when TTL = 0 A new `stunaddr_reresolve_ttl_0` parameter has been added to rtp.conf that allows control over what happens when a STUN server hostname lookup returns a TTL of 0. The values can be set as follows: - 'no': This is the historical (and current default) behavior of not doing any further lookups and continuing to use the last successful result until Asterisk is restarted or rtp.conf is reloaded. - 'yes': Use the last cached result for the current call but trigger re-resolution in the background for the benefit of future calls. If the result of the background lookup is a ttl > 0, periodic resolution will be restarted otherwise the next call will use the new cached value and will trigger a background lookup again. A new CLI command `rtp resolve stun hostname` has been added - #### app_dial: Properly handle callee hangup while sending digits. If a called channel sends progress or wink and the caller begins sending digits but the callee answers and then hangs up before digit sending can finish, the call is now answered before being disconnected. If the callee hangs up without answering, the call now continues in the dialplan. - #### Upgrade bundled pjproject to 2.17. Bundled pjproject has been upgraded to 2.17. For more information about what is included in this release, see the pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.17 - #### res_pjsip: Add per-endpoint RTP port range configuration PJSIP endpoints now support rtp_port_start and rtp_port_end options to configure a dedicated RTP port range per endpoint, overriding the global rtp.conf setting. - #### stasis_broadcast: Add optional ARI broadcast with first-claim-wins New optional modules res_stasis_broadcast.so and app_stasis_broadcast.so enable broadcasting an incoming channel to multiple ARI applications. The first application to successfully claim (via POST /ari/events/claim) wins channel control. StasisBroadcast() dialplan application initiates broadcasts. CallBroadcast and CallClaimed events notify applications. When modules are not loaded, behavior is unchanged. - #### chan_iax2: Add CHANNEL getter to retrieve auth method. CHANNEL(auth_method) can now be used to retrieve the auth method negotiated for a call on IAX2 channels. - #### res_pjsip_maintenance: Add PJSIP endpoint maintenance mode New module res_pjsip_maintenance adds runtime maintenance mode for PJSIP endpoints. Use "pjsip set maintenance " to enable or disable, and "pjsip show maintenance" to list affected endpoints. AMI actions PJSIPSetMaintenance and PJSIPShowMaintenance provide programmatic access. No configuration file changes required. ### Upgrade Notes: - #### jansson: Upgrade version to jansson 2.15.0 jansson has been upgraded to 2.15.0. For more information visit jansson Github page: https://github.com/akheron/jansson/releases/tag/v2.15.0 - #### res_pjsip: Add per-endpoint RTP port range configuration An alembic database migration has been added to add the rtp_port_start and rtp_port_end columns to the ps_endpoints table. Run "alembic upgrade head" to apply the schema change. ### Developer Notes: - #### res_pjsip: Add per-endpoint RTP port range configuration New public API: ast_rtp_instance_new_with_port_range() creates an RTP instance with a per-instance port range. ast_rtp_instance_get_port_start() and ast_rtp_instance_get_port_end() allow RTP engines to query the override. Third-party RTP engines can use these getters to support per-instance port ranges. - #### stasis_broadcast: Add optional ARI broadcast with first-claim-wins New public APIs in stasis_app_broadcast.h: stasis_app_broadcast_channel(), stasis_app_claim_channel(), stasis_app_broadcast_winner(), and stasis_app_broadcast_wait(). New ARI event types (CallBroadcast, CallClaimed) added to events.json. All code is isolated; no existing ABI modified. - #### res_pjsip_maintenance: Add PJSIP endpoint maintenance mode ast_sip_session_supplement gains a new optional callback - int (*session_create)(struct ast_sip_endpoint *endpoint, const char *destination). It is called from the global supplement list (not per-session) at the start of ast_sip_session_create_outgoing() via ast_sip_session_check_supplement_create(). Returning non-zero blocks the outgoing session. Modules that need to gate outbound SIP session creation should register a supplement with this callback set rather than hooking into chan_pjsip directly. - #### build: remove pjsua, pjsystest, Python bindings and asterisk_malloc_debug stubs from pjproject dev build The pjsua and pjsystest application binaries, the deprecated Python pjsua bindings (`_pjsua.so`), and the `asterisk_malloc_debug.c` stub implementations are no longer built or installed as part of the bundled pjproject dev mode build. The `PYTHONDEV` (python2.7-dev) build dependency is also removed. Developers who relied on the pjsua binary for Test Suite SIP simulation should use SIPp instead, which is the current Asterisk Test Suite standard. Fixes: #1840 ## Issue and Commit Detail: ### Closed Issues: - 1217: [bug]: INSERT INTO cdr query prepare statement issue on cdr_adaptive_odbc to control statement preparation manually - 1357: [bug]: MessageSend WARNING “not a valid SIP/SIPS URI” when using endpoint not URI - 1653: [bug]: Asterisk ODBC Voicemail Crash Caused by Voicemail Re-entry Loop and Unsafe BLOB Retrieval - 1736: app_queue: update_queue() may double-increment member->calls with shared_lastcall=yes (regression observed after 20.17; impacts fewestcalls routing) - 1761: func_talkdetect.c: TALK_DETECT docs wording mistake - 1762: [bug]: 100% CPU usage when entering BridgeWait after JITTERBUFFER(disabled)= - 1807: [new-feature]: translate.c: implement different types of sample frame inputs - 1812: [new-feature]: add tests/test_codec_translations.c - 1818: [bug]: func_odbc: possible use-after-free crash during reload with active calls - 1839: Crash in MDMF Caller ID parser due to signed char length field on DAHDI channels - 1840: [bug]: Asterisk fails to compile with --enable-dev-mode=yes due to INIT_RETURN undeclared in bundled pjproject Python bindings - 1855: [bug]: core reload deadlocks Asterisk (pjsip, CLI, etc.) - 1858: [bug]: DNS records with a TTL of zero are permanently cached - 1859: [bug]: res_pjsip_outbound_registration: No expires header set when triggered via CLI - 1861: [bug]: Possible heap corruption in audiohook/translate write path during bridged media - 1862: [bug]: Build fails with Building Documentation: line 210: /tmp/xmldoc.tmp.xml: Permission denied - 1865: [bug]: chan_iax2: Another code path that causes crashes on negative data lengths - 1867: [bug]: Massive [eventpoll] file-descriptor leak (hundreds of epoll fds) when TURN is enabled in rtp.conf - 1872: [bug]: Deadlock in chan_pjsip_new when endpoint set_var invokes PJSIP_HEADER - 1878: [new-feature]: chan_iax2: Allow retrieving the auth method using the CHANNEL function - 1883: [bug]: fix: stdatomic.h false positive on GCC 4.8 - 1885: [bug]: cdrel_custom :SQLite version too old: sqlite3_prepare_v3 / SQLITE_PREPARE_PERSISTENT undeclared - 1888: [improvement]: pjsip: Upgrade bundled version to pjproject 2.17 - 1892: [bug]: Build failure with bundled pjproject on OpenSSL 1.0.x: undefined reference to TLS_method and SSL_CTX_set_ciphersuites - 1894: [bug]: Outbound ARI websockets don't always clean up completely - 1896: [bug]: asterisk.c fails to compile when HAVE_LIBEDIT_IS_UNICODE isn't defined - 1901: [bug]: QUEUE_RAISE_PENALTY=rN ignored when set via queue rules - 1903: [bug]: g++ 16 no longer defines __STDC_VERSION__ causing channelstorage_cpp_map_name_id.cc to fail - 1907: [bug]: Deadlock between bridge and setting of RTP stats variables at hangup - 1910: [improvement]: Add attachable state variables to Channels and Bridges. - 1915: [bug]: app_dial: Channel not handled properly if callee disconnects while caller is sending it digits prior to answer - 1921: [bug]: Memory error in crypto_get_cert_subject when using malloc_debug - 1928: [bug]: Calling ast_softhangup with channel lock held can cause deadlock - 1931: [improvement]: jansson: Upgrade version to jansson 2.15.0 - 1936: [bug]: Calling set_variable on PJSIP channel when originating with ARI with PJSIP_HEADER can result in deadlock - 1938: [bug]: res_rtp_asterisk: Copy/paste error in ast_rtp_get_stat() - 1941: [bug]: chan_websocket doesn't handle CONTINUATION websocket frames - 1947: [bug]: chan_dahdi fails to build with gcc-16 when openr2 is installed - 1950: [bug]: app_record does not detect channel hangup during beep playback - 1952: [bug]: OpenSSL 4.0.0 - 1957: [bug]: Calendar module fails to build with libical 4.X - 1970: [bug]: Startup or shutdown segfault in res_ari_model under certain conditions with DEVMODE and persistent outbound websockets. ### Commit List: - res_ari: Add res_ari_model as an optional_module. - res ari: Add attachable states to Channels and Bridges - ARI: Added paths to get and set multiple channel variables. - res_stir_shaken: avoid direct ASN1_STRING accesses - tcptls.c: fix build with OpenSSL 4 - res_calendar: Fix build with libical 4.X - app_record: Fix hangup handling during beep playback - odbc: Don't use prepared statements for distinct SQL statements - abstract_jb.c: Remove timerfd from channel when disabling jitter buffer - res_pjsip: Don't allow a leading period when wildcard matching - Ensure channel locks aren't held while calling ast_set_variables. - app_queue: fix double increment of member->calls with shared_lastcall - chan_dahdi: Fix set but not used in mfcr2_show_links_of(). - tests: add tests/test_codec_translations.c - install_prereq: Add a 'minimal' mode for basic build dependencies - chan_websocket: Handle incoming CONTINUATION frames. - res_rtp_asterisk: Fix incorrect reference in ast_rtp_get_stat(). - jansson: Upgrade version to jansson 2.15.0 - channel.c: Move setting RTP stats from ast_softhangup to ast_ari_channels_hangup. - res_rtp_asterisk: Add option to control stun host resolution when TTL = 0 - pjsip_configuration: Show actual dtls_verify config. - app_dial: Properly handle callee hangup while sending digits. - res_pjsip_messaging: Update To URI only if it is a SIP(S) URI - Upgrade bundled pjproject to 2.17. - res_stir_shaken: fix memory free crash when Asterisk is built with malloc_debug - manager: Eliminate unnecessary code, simplify sessions in stasis callbacks - res_stasis/resource_bridges: Split bridge playback control and wrapper cleanup - res_pjsip_outbound_publish.c: Add more verbose documentation for outbound_proxy usage - channel.c: Don't lock the channel in ast_softhangup while setting rtp instance vars - chan_pjsip: Fix deadlock when endpoint set_var uses PJSIP_HEADER - res_pjsip: Add per-endpoint RTP port range configuration - app_queue: Fix raise_respect_min lost in copy_rules() breaking rN queue rules - app_voicemail_odbc: fix msgnum race and crash on failed STORE - ari_websockets: Fix two issues in the cleanup of outbound websockets. - compat.h: Ensure check for `__STDC_VERSION__` is not attempted for c++. - pjproject: Backport fix for OpenSSL < 1.1.0 build failure in ssl_sock_ossl.c - asterisk.c: Fix #if HAVE_LIBEDIT_IS_UNICODE. - cdrel_custom: fix SQLite compatibility for versions < 3.20.0 - translate.c: implement different sample_types for translation computation. - stasis_broadcast: Add optional ARI broadcast with first-claim-wins - res_audiosocket: Tolerate non-audio frame types - pbx_functions: Save module pointer before calling read and write callbacks. - chan_iax2: Add CHANNEL getter to retrieve auth method. - fix: backport pjproject stdatomic.h GCC 4.8 build failure patch - res_rtp_asterisk: Destroy ioqueue in rtp_ioqueue_thread_destroy. - res_pjsip_maintenance: Add PJSIP endpoint maintenance mode - chan_iax2: Add another check to abort frame handling if datalen < 0. - res_pjsip_outbound_registration: only update the Expires header if the value has changed - func_talkdetect.c: Clarify dsp_talking_threshold documentation. - make_xml_documentation: Remove temporary file on script exit. - res_pjsip_config_wizard: Trigger reloads from a pjsip servant thread - build: remove pjsua, pjsystest, Python bindings and asterisk_malloc_debug stubs from pjproject dev build - callerid: fix signed char causing crash in MDMF parser @ text @# $NetBSD: Makefile,v 1.8 2026/05/14 16:40:32 ryoon Exp $ # # NOTE: when updating this package, there are two places that sound # tarballs need to be checked; look in ${WRKSRC}/sounds/Makefile # to find out the current sound file versions # Also look in ${WRKSRC}/third-party/versions.mak for pjproject DISTNAME= asterisk-23.4.0 CATEGORIES= comms net audio MASTER_SITES= https://downloads.asterisk.org/pub/telephony/asterisk/ MASTER_SITES+= https://downloads.asterisk.org/pub/telephony/asterisk/old-releases/ MASTER_SITES+= https://downloads.asterisk.org/pub/telephony/sounds/releases/ DIST_SUBDIR= ${PKGNAME_NOREV} DISTFILES= ${DEFAULT_DISTFILES} COMMENT= The Asterisk Software PBX LICENSE= gnu-gpl-v2 EXTRACT_ONLY= ${DISTNAME}.tar.gz MAINTAINER= jnemeth@@NetBSD.org HOMEPAGE= https://www.asterisk.org/ MAKE_JOBS_SAFE= NO # known to have issues on i386, block the package until the bug is fixed BROKEN_ON_PLATFORM= NetBSD-*-i386 CONFLICTS+= asterisk-sounds-extra-[0-9]* .include "../../mk/bsd.prefs.mk" USE_TOOLS+= bison gmake gsed perl:run pkg-config tar bash:run USE_LANGUAGES= c c++ REPLACE_BASH+= contrib/scripts/astversion REPLACE_BASH+= contrib/scripts/ast_coredumper REPLACE_BASH+= contrib/scripts/ast_logescalator REPLACE_BASH+= contrib/scripts/ast_loggrabber REPLACE_PERL+= agi/DialAnMp3.agi agi/agi-test.agi REPLACE_PERL+= agi/fastagi-test agi/jukebox.agi agi/numeralize REPLACE_PERL+= contrib/scripts/vmail.cgi REPLACE_PYTHON+= contrib/scripts/reflocks.py contrib/scripts/refstats.py CHECK_INTERPRETER_SKIP+= libdata/asterisk/scripts/refcounter.py GNU_CONFIGURE= yes CONFIGURE_ARGS+= --datarootdir=${PREFIX}/libdata CONFIGURE_ARGS+= --sysconfdir=${PKG_SYSCONFDIR} CONFIGURE_ARGS+= --without-gtk2 # XXX remove when lang/lua gets builtin.mk CONFIGURE_ARGS+= --without-lua CONFIGURE_ARGS+= --without-unbound CONFIGURE_ENV+= ac_cv_path_XMLSTARLET=":" # XXX uncomment below to debug build issues #CONFIGURE_ENV+= PJPROJECT_CONFIGURE_OPTS="NOISY_BUILD=yes" .if ${OPSYS} == "NetBSD" && ${OPSYS_VERSION} < 069900 CONFIGURE_ARGS+= --without-srtp .else CONFIGURE_ARGS+= --with-srtp PLIST.srtp= YES .include "../../comms/srtp/buildlink3.mk" .endif INSTALL_TARGET= install samples INSTALLATION_DIRS+= lib/pkgconfig share/doc/${PKGBASE} INSTALLATION_DIRS+= share/examples/asterisk share/examples/rc.d INSTALLATION_DIRS+= ${ASTDATADIR}/sounds/en ${ASTDATADIR}/moh BUILD_DEFS+= VARBASE ASTERISK_USER?= asterisk ASTERISK_GROUP?= asterisk PKG_GROUPS= ${ASTERISK_GROUP} PKG_USERS= ${ASTERISK_USER}:${ASTERISK_GROUP} PKG_GECOS.${ASTERISK_USER}= Asterisk PBX PKG_GROUPS_VARS= ASTERISK_GROUP PKG_USERS_VARS= ASTERISK_USER FILES_SUBST+= ASTERISK_USER=${ASTERISK_USER} FILES_SUBST+= ASTERISK_GROUP=${ASTERISK_GROUP} MESSAGE_SUBST+= ASTERISK_USER=${ASTERISK_USER} MESSAGE_SUBST+= ASTERISK_GROUP=${ASTERISK_GROUP} # Various path settings for Asterisk PKG_SYSCONFSUBDIR= asterisk PKG_SYSCONFDIR_PERMS= ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 ASTETCDIR= ${PKG_SYSCONFDIR} ASTEXAMPLEDIR= ${PREFIX}/share/examples/asterisk ASTDBDIR= ${VARBASE}/db/asterisk ASTSPOOLDIR= ${VARBASE}/spool/asterisk ASTLOGDIR= ${VARBASE}/log/asterisk MESSAGE_SUBST+= ASTDBDIR=${ASTDBDIR} MESSAGE_SUBST+= ASTSPOOLDIR=${ASTSPOOLDIR} MESSAGE_SUBST+= ASTLOGDIR=${ASTLOGDIR} ASTVARRUNDIR= ${VARBASE}/run/asterisk FILES_SUBST+= ASTVARRUNDIR=${ASTVARRUNDIR} ASTDATADIR= ${PREFIX}/libdata/asterisk MAKE_FLAGS+= ASTLIBDIR=${PREFIX}/lib/asterisk MAKE_FLAGS+= ASTVARLIBDIR=${ASTDATADIR} MAKE_FLAGS+= ASTKEYDIR=${ASTDATADIR} MAKE_FLAGS+= ASTDATADIR=${ASTDATADIR} MAKE_FLAGS+= ASTSPOOLDIR=${ASTSPOOLDIR} MAKE_FLAGS+= ASTLOGDIR=${ASTLOGDIR} MAKE_FLAGS+= ASTHEADERDIR=${PREFIX}/include/asterisk MAKE_FLAGS+= ASTCONFPATH=${ASTETCDIR}/asterisk.conf MAKE_FLAGS+= ASTBINDIR=${PREFIX}/bin MAKE_FLAGS+= ASTSBINDIR=${PREFIX}/sbin MAKE_FLAGS+= ASTVARRUNDIR=${ASTVARRUNDIR} MAKE_FLAGS+= ASTMANDIR=${PREFIX}/${PKGMANDIR} MAKE_FLAGS+= ASTETCDIR=${ASTETCDIR} MAKE_FLAGS+= ASTDBDIR=${ASTDBDIR} MAKE_FLAGS+= AGI_DIR=${PREFIX}/libexec/agi-bin MAKE_FLAGS+= ASTEXAMPLEDIR=${ASTEXAMPLEDIR} MAKE_FLAGS+= WRKSRC=${WRKSRC} MAKE_FLAGS+= LDOPTS=${LDFLAGS:M*:Q} MAKE_FLAGS+= HTTP_DOCSDIR=${PREFIX}/share/httpd/htdocs MAKE_FLAGS+= HTTP_CGIDIR=${PREFIX}/libexec/cgi-bin MAKE_FLAGS+= OPTIMIZE=-O3 # uncomment the following to debug build problems #MAKE_FLAGS+= NOISY_BUILD=yes .if ${MACHINE_PLATFORM:MSunOS-*-i386} BUILDLINK_TRANSFORM+= rm:-march=i386 .endif PLIST_VARS+= kqueue .if defined(OPSYS_HAS_KQUEUE) PLIST.kqueue= yes .endif PLIST_VARS+= timerfd .if defined(OPSYS_HAS_TIMERFD) PLIST.timerfd= yes .else CONFIGURE_ARGS+= --without-timerfd .endif .include "options.mk" # check sounds/Makefile for current version when upgrading package DISTFILES+= asterisk-extra-sounds-en-gsm-1.5.2.tar.gz # pjproject PJPROJ_VERSION= 2.17 SITES.pjproject-${PJPROJ_VERSION}.tar.bz2= \ -https://raw.githubusercontent.com/asterisk/third-party/master/pjproject/${PJPROJ_VERSION}/pjproject-${PJPROJ_VERSION}.tar.bz2 SITES.pjproject-${PJPROJ_VERSION}.md5= \ -https://raw.githubusercontent.com/asterisk/third-party/master/pjproject/${PJPROJ_VERSION}/MD5SUM.TXT DISTFILES+= pjproject-${PJPROJ_VERSION}.tar.bz2 pjproject-${PJPROJ_VERSION}.md5 ALL_ENV+= EXTERNALS_CACHE_DIR=${DISTDIR}/${DIST_SUBDIR} # Override default paths in config files SUBST_CLASSES+= configs SUBST_STAGE.configs= pre-configure SUBST_FILES.configs= configs/samples/cli_aliases.conf.sample SUBST_FILES.configs+= configs/samples/festival.conf.sample SUBST_FILES.configs+= configs/samples/http.conf.sample SUBST_FILES.configs+= configs/samples/iax.conf.sample SUBST_FILES.configs+= configs/samples/musiconhold.conf.sample SUBST_FILES.configs+= configs/samples/phoneprov.conf.sample SUBST_SED.configs+= -e 's|doc/|${PREFIX}/share/doc/${PKGBASE}/|' SUBST_SED.configs+= -e 's|/etc/asterisk|${ASTETCDIR}|' SUBST_SED.configs+= -e 's|/var/lib/asterisk|${ASTVARLIBDIR}|' SUBST_SED.configs+= -e "s|/usr/local/man|${ASTMANDIR}|" SUBST_SED.configs+= -e "s|/usr/local|${PREFIX}|" SUBST_SED.configs+= -e "s|/var|${VARBASE}|" # XXX gross hack, remove when item is fixed SUBST_CLASSES.NetBSD+= autoconfig SUBST_STAGE.autoconfig= post-configure SUBST_FILES.autoconfig= include/asterisk/autoconfig.h SUBST_SED.autoconfig= -e "s|^\#define HAVE_GCC_ATOMICS 1|\#undef HAVE_GCC_ATOMICS|" SUBST_SED.autoconfig+= -e "s|^\#define HAVE_GETHOSTBYNAME_R_6 1|\#undef HAVE_GETHOSTBYNAME_R_6|" # XXX gross hack, IP_PKTINFO in NetBSD isn't compatible with anything else SUBST_CLASSES.NetBSD+= pktinfo SUBST_STAGE.pktinfo= post-configure SUBST_FILES.pktinfo= include/asterisk/autoconfig.h SUBST_SED.pktinfo= -e "s|^\#define HAVE_PKTINFO 1|\#undef HAVE_PKTINFO|" RCD_SCRIPTS= asterisk OWN_DIRS_PERMS+= ${ASTDBDIR} ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTSPOOLDIR} ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTSPOOLDIR}/dictate ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTSPOOLDIR}/meetme ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTSPOOLDIR}/monitor ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTSPOOLDIR}/outgoing ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTSPOOLDIR}/system ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTSPOOLDIR}/tmp ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTSPOOLDIR}/voicemail ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTSPOOLDIR}/voicemail/default ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTSPOOLDIR}/voicemail/default/1234 ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTSPOOLDIR}/voicemail/default/1234/INBOX ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTSPOOLDIR}/voicemail/default/1234/en ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 SPECIAL_PERMS+= ${ASTSPOOLDIR}/voicemail/default/1234/en/busy.gsm ${ASTERISK_USER} ${ASTERISK_GROUP} 0644 SPECIAL_PERMS+= ${ASTSPOOLDIR}/voicemail/default/1234/en/unavail.gsm ${ASTERISK_USER} ${ASTERISK_GROUP} 0644 OWN_DIRS_PERMS+= ${ASTSPOOLDIR}/voicemail/default/INBOX ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTSPOOLDIR}/voicemail/default/en ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTLOGDIR} ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTLOGDIR}/cdr-csv ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 OWN_DIRS_PERMS+= ${ASTLOGDIR}/cdr-custom ${ASTERISK_USER} ${ASTERISK_GROUP} 0755 CONF_FILES_PERMS= # empty .for f in asterisk.conf extensions.conf CONF_FILES_PERMS+= ${ASTEXAMPLEDIR}/${f:Q} ${PKG_SYSCONFDIR}/${f:Q} ${ASTERISK_USER} ${ASTERISK_GROUP} 0644 .endfor .if !empty(PKG_OPTIONS:Masterisk-config) # if we put all the files in $CONF_FILES, the message is _way_ too long. . for f in acl.conf adsi.conf aeap.conf agents.conf alarmreceiver.conf \ amd.conf app_skel.conf ari.conf ast_debug_tools.conf \ asterisk.adsi calendar.conf ccss.conf cdr.conf \ cdr_adaptive_odbc.conf cdr_beanstalkd.conf cdr_custom.conf \ cdr_manager.conf cdr_odbc.conf cdr_pgsql.conf \ cdr_sqlite3_custom.conf cdr_tds.conf cel.conf \ cel_beanstalkd.conf cel_custom.conf cel_odbc.conf \ cel_pgsql.conf cel_sqlite3_custom.conf cel_tds.conf \ chan_dahdi.conf chan_mobile.conf chan_websocket.conf cli.conf \ cli_aliases.conf cli_permissions.conf codecs.conf \ confbridge.conf config_test.conf console.conf dbsep.conf \ dnsmgr.conf dsp.conf dundi.conf enum.conf extconfig.conf \ extensions.ael extensions.conf extensions.lua \ extensions_minivm.conf features.conf festival.conf \ followme.conf func_odbc.conf geolocation.conf hep.conf \ http.conf iax.conf iaxprov.conf indications.conf logger.conf \ manager.conf meetme.conf minivm.conf modules.conf motif.conf \ musiconhold.conf ooh323.conf phoneprov.conf \ phoneprov_users.conf pjproject.conf pjsip.conf \ pjsip_notify.conf pjsip_wizard.conf prometheus.conf \ queuerules.conf queues.conf res_config_mysql.conf \ res_config_odbc.conf res_config_sqlite3.conf res_corosync.conf \ res_curl.conf res_fax.conf res_http_media_cache.conf \ res_ldap.conf res_odbc.conf res_parking.conf res_pgsql.conf \ res_snmp.conf res_stun_monitor.conf resolver_unbound.conf \ rtp.conf say.conf sla.conf smdi.conf sorcery.conf ss7.timers \ stasis.conf statsd.conf stir_shaken.conf telcordia-1.adsi \ test_sorcery.conf udptl.conf unistim.conf voicemail.conf \ websocket_client.conf xmpp.conf CONF_FILES_PERMS+= ${ASTEXAMPLEDIR}/${f:Q} ${PKG_SYSCONFDIR}/${f:Q} ${ASTERISK_USER} ${ASTERISK_GROUP} 0644 . endfor .endif PTHREAD_OPTS+= require native .if ${OPSYS} == "Darwin" && exists(/usr/include/sys/poll.h) post-patch: ${ECHO} "#include " > ${WRKSRC}/include/asterisk/poll-compat.h .endif post-install: # check sounds directory for current versions when upgrading package ${TAR} xzf ${WRKSRC}/sounds/asterisk-core-sounds-en-gsm-1.6.1.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en ${TAR} xzf ${WRKSRC}/sounds/asterisk-moh-opsound-wav-2.03.tar.gz -C ${DESTDIR}${ASTDATADIR}/moh ${TAR} xzf ${DISTDIR}/${DIST_SUBDIR}/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en ${INSTALL_DATA} ${WRKSRC}/BUGS ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/COPYING ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/CREDITS ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-23.0.0.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-23.0.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-23.1.0.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-23.1.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-23.2.0.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-23.2.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-23.2.1.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-23.2.1.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-23.2.2.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-23.2.2.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-23.3.0.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-23.3.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-23.4.0.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-23.4.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/LICENSE ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/README-SERIOUSLY.bestpractices.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/README.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/SECURITY.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/Zaptel-to-DAHDI.txt ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/doc/IAX2-security.pdf ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/doc/IAX2-security.txt ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} ${INSTALL_DATA} ${WRKSRC}/doc/README.txt ${DESTDIR}${PREFIX}/share/doc/${PKGBASE} .if ${OPSYS} != "Darwin" && ${PKGSRC_COMPILER:Mclang} .include "../../lang/libBlocksRuntime/buildlink3.mk" LDFLAGS+= -L${PREFIX}/lib .endif .include "../../databases/sqlite3/buildlink3.mk" .include "../../devel/editline/buildlink3.mk" .include "../../devel/libuuid/buildlink3.mk" .include "../../devel/zlib/buildlink3.mk" .include "../../lang/python/application.mk" .include "../../textproc/jansson/buildlink3.mk" .include "../../security/openssl/buildlink3.mk" .include "../../textproc/libxml2/buildlink3.mk" .include "../../www/curl/buildlink3.mk" .include "../../mk/curses.buildlink3.mk" .include "../../mk/pthread.buildlink3.mk" .include "../../mk/termcap.buildlink3.mk" .include "../../mk/bsd.pkg.mk" @ 1.8 log @*: Recursive revbump from security/nettle-4.0 @ text @d1 1 a1 1 # $NetBSD: Makefile,v 1.7 2026/04/13 03:47:11 jnemeth Exp $ d8 1 a8 2 DISTNAME= asterisk-23.3.0 PKGREVISION= 1 d143 1 a143 1 PJPROJ_VERSION= 2.16 d269 2 @ 1.7 log @Update to Asterisk 23.3.0: ## Change Log for Release asterisk-23.3.0 ### Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.3.0.html) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.2...23.3.0) ### Summary: - Commits: 50 - Commit Authors: 21 - Issues Resolved: 34 - Security Advisories Resolved: 0 ### User Notes: - #### acl: Add ACL support to http and ari A new section, type=restriction has been added to http.conf to allow an uri prefix based acl to be configured. See http.conf.sample for examples and more information. The user section of ari.conf can now contain an acl configuration to restrict users access. See ari.conf.sample for examples and more information - #### res_rtp_asterisk.c: Fix DTLS packet drop when TURN loopback re-injection occurs before ICE candidate check WebRTC calls using TURN configured in rtp.conf (turnaddr, turnusername, turnpassword) will now correctly complete DTLS/SRTP negotiation. Previously all DTLS packets were silently dropped due to the loopback re-injection address not being in the ICE active candidate list. - #### docs: Add "Provided-by" to doc XML and CLI output. The CLI help for applications, functions, manager commands and manager events now shows the module that provides its functionality. - #### CDR/CEL Custom Performance Improvements Significant performance improvements have been made to the cdr_custom, cdr_sqlite3_custom, cel_custom and cel_sqlite3_custom modules. See the new sample config files for those modules to see how to benefit from them. - #### chan_websocket: Add media direction. WebSocket now supports media direction, allowing for unidirectional media. This is done from the perspective of the application and can be set via channel origination, external media, or commands sent from the application. Check out https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket/ for more. - #### app_queue: Add 'prio' setting to the 'force_longest_waiting_caller' option The 'force_longest_waiting_caller' option now supports a 'prio' setting. When set to 'prio', calls are offered by priority first, then by wait time. - #### Upgrade bundled pjproject to 2.16. Bundled pjproject has been upgraded to 2.16. For more information on what all is included in this change, check out the pjproject Github page: https://github.com/pjsip/pjproject/releases - #### res_pjsip_header_funcs: Add new PJSIP_INHERITABLE_HEADER dialplan function A new PJSIP_HEADER option has been added that allows inheriting pjsip headers from the inbound to the outbound bridged channel. Example- same => n,Set(PJSIP_INHERITABLE_HEADER(add,X-custom-1)=alpha) will add X-custom-1: alpha to the outbound pjsip channel INVITE upon Dial. - #### app_queue: Fix rN raise_penalty ignoring min_penalty in calc_metric Fixes an issue where QUEUE_RAISE_PENALTY=rN could raise a member’s penalty below QUEUE_MIN_PENALTY during member selection. This could allow members intended to be excluded to be selected. The queue now consistently respects the minimum penalty when raising penalties, aligning member selection behavior with queue empty checks and documented rN semantics. ## Issue and Commit Detail: ### Closed Issues: - 449: [bug]: PJSIP confuses media address after INVITE requiring authentication - 566: [bug]: core: SIGSEGV on DTMF when no timing modules loaded - 1356: [bug]: MESSAGE requests should not contain a Contact header - 1524: [bug]: PJSIP if sdp_session is blank the initial INVITE doesn't attach an SDP offer, worked in chan_sip - 1611: [bug]: asterisk deadlocked on start sometimes - 1612: [improvement]: pjsip: Upgrade bundled version to pjproject 2.16 - 1637: [improvement]: force_longest_waiting_caller should also consider caller priority - 1641: [bug]: res_pjsip_config_wizard: Endpoints fail to update when Named ACLs change after reload - 1651: [bug]: Asterisk crashes with munmap_chunk() when using sorcery realtime for PJSIP registration objects - 1657: [bug]: Wrong dtmf payload is used when inbound invite contains 8K and 16K, and outgoing leg is using G722 and SRTP - 1670: [new-feature]: Add new option to PJSIP_HEADER to pass headers from the inbound to outbound channel. - 1691: [bug]: force_longest_waiting_caller stops offering calls if a call joins at the first position - 1703: [bug]: res_pjsip_pubsub: ao2 reference leak of subscription tree in ast_sip_subscription - 1707: [bug]: chan_iax2: Crash when processing video frames with negative length - 1716: [bug]: Ghost call when UAC didn't respond with 487 for a cancel request from server even after original call hangup. - 1724: [improvement]: say.c - added language support for pashto and dari - 1730: [bug]: CPP channel storage get_by_name_prefix does not check prefix match - 1755: [bug]: app_dial, utils.h: Compilation failure with -Wold-style-declaration and -Wdiscarded-qualifiers - 1781: [bug]: More discarded-qualifiers errors with gcc 15.2.1 - 1783: [bug]: Several unused-but-set-variable warnings with gcc 16 - 1785: [bug]: chan_websocket doesn’t work with genericplc and transcoding - 1786: [bug]: chan_dahdi: A few more discarded-qualifiers errors not caught previously - 1795: [bug]: DTLS packets dropped when TURN configured in rtp.conf due to loopback re-injection occurring before ICE candidate source check - 1797: [bug]: Potential logic issue in translated frame write loop (main/file.c) - 1802: [improvement]: app_dial: Channel name should be included in warnings during wait_for_answer - 1804: [new-feature]: dsp.c: Add support for R2 signaling - 1814: [bug]: A pjsip transport with an invalid config can cause issues with other transports - 1816: [bug]: ARI: RTPAUDIO channel vars aren't set if call hung up by ARI. - 1819: [bug]: When a 302 is received from a UAS, the cause and tech_cause codes set on the channel are incorrect. - 1831: [bug]:raise_exception() and EXCEPTION() read use channel datastores without holding ast_channel_lock - 1833: [bug]: Address security vulnerabilities in pjproject - 1844: [bug]: cdrel_custom isn't respecting the default time format for CEL records - 1845: [bug]:res_cdrel_custom produces wrong float timestamps - 1852: [bug]: res_cdrel_custom: connection to the sqlite3 database closes from time to time ### Commit List: - res_cdrel_custom: do not free config when no new config was loaded - res_cdrel_custom: Resolve several formatting issues. - res_pjsip: Address pjproject security vulnerabilities - pbx: Hold channel lock for exception datastore access - xmldoc.c: Fix memory leaks in handling of provided_by. - SECURITY.md: Update with additional instructions. - res_audiosocket: Fix header read loop to use correct buffer offset - manager.c : Fix CLI event display - chan_pjsip: Set correct cause codes for non-2XX responses. - res_pjsip_config_wizard: Force reload on Named ACL change events - rtp: Set RTPAUDIOQOS variables when ast_softhangup is called. - channel: Prevent crash during DTMF emulation when no timing module is loaded - res_pjsip: Remove temp transport state when a transport fails to load. - res_pjsip_messaging: Remove Contact header from out-of-dialog MESSAGE as per RFC3428 - acl: Add ACL support to http and ari - res_rtp_asterisk.c: Fix DTLS packet drop when TURN loopback re-injection occurs before ICE candidate check - dsp.c: Add support for detecting R2 signaling tones. - app_dial: Include channel name in warnings during wait_for_answer. - main/file: fix translated-frame write loop to use current frame - docs: Add "Provided-by" to doc XML and CLI output. - chan_websocket_doc.xml: Add d(media_direction) option. - resource_channels.c: Fix validation response for externalMedia with AudioSockets - CDR/CEL Custom Performance Improvements - chan_websocket: Remove silence generation and frame padding. - chan_websocket: Add media direction. - fix: Add macOS (Darwin) compatibility for building Asterisk - astconfigparser.py: Fix regex pattern error by properly escaping string - res_rtp_asterisk: use correct sample rate lookup to account for g722 - res_pjsip_outbound_registration.c: Prevent crash if load_module() fails - pjsip_configuration: Ensure s= and o= lines in SDP are never empty - res_pjsip_session: Make sure NAT hook runs when packet is retransmitted for whatever reason. - chan_dahdi: Fix discarded-qualifiers errors. - build: Fix unused-but-set-variable warnings with gcc 16. - build: Fix another GCC discarded-qualifiers const error. - chan_iax2: Fix crash due to negative length frame lengths. - build: Fix GCC discarded-qualifiers const errors. - endpoints: Allow access to latest snapshot directly. - app_dial, utils.h: Avoid old style declaration and discarded qualifier. - app_queue: Add 'prio' setting to the 'force_longest_waiting_caller' option - Upgrade bundled pjproject to 2.16. - res_pjsip_header_funcs: Add new PJSIP_INHERITABLE_HEADER dialplan function - app_queue: Queue Timing Parity with Dial() and Accurate Wait Metrics - stasis.c: Fix deadlock in stasis_topic_pool_get_topic during module load - app_queue: Fix rN raise_penalty ignoring min_penalty in calc_metric - app_queue: Only compare calls at 1st position across queues when forcing longest waiting caller. - channelstorage_cpp_map_name_id: Fix get_by_name_prefix prefix match - app_amd: Remove errant space in documentation for totalAnalysisTime. - say.c: added language support for pashto and dari - res_pjsip_session.c: Prevent INVITE failover when session is cancelled - res_pjsip_pubsub: Fix ao2 reference leak of subscription tree in ast_sip_subscription @ text @d1 1 a1 1 # $NetBSD: Makefile,v 1.6 2026/04/10 08:41:35 wiz Exp $ d9 1 @ 1.6 log @*: remove OWNER definition OWNER, when it was introduced, was to protect packages deep in the infrastructure by emphasizing that they should not be touched by non-MAINTAINERs. No infrastructure package still sets OWNER. Note: non-trivial change to packages should be passed by MAINTAINERs. As discussed on tech-pkg. @ text @d1 1 a1 1 # $NetBSD: Makefile,v 1.5 2026/02/16 04:11:13 jnemeth Exp $ d8 1 a8 1 DISTNAME= asterisk-23.2.2 d143 1 a143 1 PJPROJ_VERSION= 2.15.1 d267 2 @ 1.5 log @update to Asterisk 23.2.2: ----- 23.2.2 ----- ## Change Log for Release asterisk-23.2.2 ### Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.2.html) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.1...23.2.2) ### Summary: - Commits: 4 - Commit Authors: 2 - Issues Resolved: 0 - Security Advisories Resolved: 4 - [GHSA-85x7-54wr-vh42](https://github.com/asterisk/asterisk/security/advisories/GHSA-85x7-54wr-vh42): Asterisk xml.c uses unsafe XML_PARSE_NOENT leading to potential XXE Injection - [GHSA-rvch-3jmx-3jf3](https://github.com/asterisk/asterisk/security/advisories/GHSA-rvch-3jmx-3jf3): ast_coredumper running as root sources ast_debug_tools.conf from /etc/asterisk; potentially leading to privilege escalation - [GHSA-v6hp-wh3r-cwxh](https://github.com/asterisk/asterisk/security/advisories/GHSA-v6hp-wh3r-cwxh): The Asterisk embedded web server's /httpstatus page echos user supplied values(cookie and query string) without sanitization - [GHSA-xpc6-x892-v83c](https://github.com/asterisk/asterisk/security/advisories/GHSA-xpc6-x892-v83c): ast_coredumper runs as root, and writes gdb init file to world writeable folder; leading to potential privilege escalation ### User Notes: - #### ast_coredumper: check ast_debug_tools.conf permissions ast_debug_tools.conf must be owned by root and not be writable by other users or groups to be used by ast_coredumper or by ast_logescalator or ast_loggrabber when run as root. ### Upgrade Notes: - #### http.c: Change httpstatus to default disabled and sanitize output. To prevent possible security issues, the `/httpstatus` page served by the internal web server is now disabled by default. To explicitly enable it, set `enable_status=yes` in http.conf. ----- 23.2.1 ----- ## Change Log for Release asterisk-23.2.1 ### Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.1.html) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.2.0...23.2.1) ### Summary: - Commits: 1 - Commit Authors: 1 - Issues Resolved: 1 - Security Advisories Resolved: 0 ## Issue and Commit Detail: ### Closed Issues: - 1739: [bug]: Regression in 23.2.0 with regard to parsing fractional numbers when system locale is non-standard ----- 23.2.0 ----- ## Change Log for Release asterisk-23.2.0 ### Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.2.0.html) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0...23.2.0) ### Summary: - Commits: 58 - Commit Authors: 20 - Issues Resolved: 41 - Security Advisories Resolved: 0 ### User Notes: - #### chan_websocket.conf.sample: Fix category name. The category name in the chan_websocket.conf.sample file was incorrect. It should be "global" instead of "general". - #### cli.c: Allow 'channel request hangup' to accept patterns. The 'channel request hangup' CLI command now accepts multiple channel names, POSIX Extended Regular Expressions, glob-like patterns, or a combination of all of them. See the CLI command 'core show help channel request hangup' for full details. - #### res_sorcery_memory_cache: Reduce cache lock time for sorcery memory cache populate command The AMI command sorcery memory cache populate will now return an error if there is an internal error performing the populate. The CLI command will display an error in this case as well. - #### res_geolocation: Fix multiple issues with XML generation. Geolocation: Two new optional profile parameters have been added. * `pidf_element_id` which sets the value of the `id` attribute on the top-level PIDF-LO `device`, `person` or `tuple` elements. * `device_id` which sets the content of the `` element. Both parameters can include channel variables. - #### res_pjsip_messaging: Add support for following 3xx redirects A new pjsip endpoint option follow_redirect_methods was added. This option is a comma-delimited, case-insensitive list of SIP methods for which SIP 3XX redirect responses are followed. An alembic upgrade script has been added for adding this new option to the Asterisk database. - #### taskprocessors: Improve logging and add new cli options New CLI command has been added - core show taskprocessor name - #### ccss: Add option to ccss.conf to globally disable it. A new "enabled" parameter has been added to ccss.conf. It defaults to "yes" to preserve backwards compatibility but CCSS is rarely used so setting "enabled = no" in the "general" section can save some unneeded channel locking operations and log message spam. Disabling ccss will also prevent the func_callcompletion and chan_dahdi modules from loading. - #### Makefile: Add module-list-* targets. Try "make module-list-deprecated" to see what modules are on their way out the door. - #### app_mixmonitor: Add 's' (skip) option to delay recording. This change introduces a new 's()' (skip) option to the MixMonitor application. Example: MixMonitor(${UNIQUEID}.wav,s(3)) This skips recording for the first 3 seconds before writing audio to the file. Existing MixMonitor behavior remains unchanged when the 's' option is not used. - #### app_queue.c: Only announce to head caller if announce_to_first_user When announce_to_first_user is false, no announcements are played to the head caller ### Upgrade Notes: - #### res_geolocation: Fix multiple issues with XML generation. Geolocation: In order to correct bugs in both code and documentation, the following changes to the parameters for GML geolocation locations are now in effect: * The documented but unimplemented `crs` (coordinate reference system) element has been added to the location_info parameter that indicates whether the `2d` or `3d` reference system is to be used. If the crs isn't valid for the shape specified, an error will be generated. The default depends on the shape specified. * The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs. If crs isn't specified, it will default to `2d` for these shapes. The Sphere, Ellipsoid and Prism shapes MUST use a `3d` crs. If crs isn't specified, it will default to `3d` for these shapes. The Point and Polygon shapes may use either crs. The default crs is `2d` however so if `3d` positions are used, the crs must be explicitly set to `3d`. * The `geoloc show gml_shape_defs` CLI command has been updated to show which coordinate reference systems are valid for each shape. * The `pos3d` element has been removed in favor of allowing the `pos` element to include altitude if the crs is `3d`. The number of values in the `pos` element MUST be 2 if the crs is `2d` and 3 if the crs is `3d`. An error will be generated for any other combination. * The angle unit-of-measure for shapes that use angles should now be included in the respective parameter. The default is `degrees`. There were some inconsistent references to `orientation_uom` in some documentation but that parameter never worked and is now removed. See examples below. Examples... ``` location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20" location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620" location_info = shape="Point", pos="39.0 -105.0" location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20" semiMinorAxis="10", verticalAxis="0", orientation="25 degrees" pidf_element_id = ${CHANNEL(name)}-${EXTEN} device_id = mac:001122334455 Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN}) ``` - #### pjsip: Move from threadpool to taskpool The threadpool_* options in pjsip.conf have now been deprecated though they continue to be read and used. They have been replaced with taskpool options that give greater control over the underlying taskpool used for PJSIP. An alembic upgrade script has been added to add these options to realtime as well. - #### app_directed_pickup.c: Change some log messages from NOTICE to VERBOSE. In an effort to reduce log spam, two normal progress "pickup attempted" log messages from app_directed_pickup have been changed from NOTICE to VERBOSE(3). This puts them on par with other normal dialplan progress messages. ### Developer Notes: - #### ccss: Add option to ccss.conf to globally disable it. A new API ast_is_cc_enabled() has been added. It should be used to ensure that CCSS is enabled before making any other ast_cc_* calls. - #### chan_websocket: Add ability to place a MARK in the media stream. Apps can now send a `MARK_MEDIA` command with an optional `correlation_id` parameter to chan_websocket which will be placed in the media frame queue. When that frame is dequeued after all intervening media has been played to the core, chan_websocket will send a `MEDIA_MARK_PROCESSED` event to the app with the same correlation_id (if any). - #### chan_websocket: Add capability for JSON control messages and events. The chan_websocket plain-text control and event messages are now deprecated (but remain the default) in favor of JSON formatted messages. See https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket for more information. A "transport_data" parameter has been added to the @ text @d1 1 a1 1 # $NetBSD: Makefile,v 1.4 2026/02/06 10:04:21 wiz Exp $ d20 1 a20 1 OWNER= jnemeth@@NetBSD.org @ 1.4 log @*: recursive bump for nettle 4.0 shlib major bump @ text @d1 1 a1 1 # $NetBSD: Makefile,v 1.3 2026/01/07 08:46:04 wiz Exp $ d8 1 a8 2 DISTNAME= asterisk-23.1.0 PKGREVISION= 2 a179 3 post-extract: mv ${WRKSRC}/configs/samples/phoneprov_users.conf ${WRKSRC}/configs/samples/phoneprov_users.conf.sample d217 12 a228 11 chan_dahdi.conf chan_mobile.conf cli.conf cli_aliases.conf \ cli_permissions.conf codecs.conf confbridge.conf \ config_test.conf console.conf dbsep.conf dnsmgr.conf dsp.conf \ dundi.conf enum.conf extconfig.conf extensions.ael \ extensions.conf extensions.lua extensions_minivm.conf \ features.conf festival.conf followme.conf func_odbc.conf \ geolocation.conf hep.conf http.conf iax.conf iaxprov.conf \ indications.conf logger.conf manager.conf meetme.conf \ minivm.conf modules.conf motif.conf musiconhold.conf \ ooh323.conf phoneprov.conf phoneprov_users.conf pjproject.conf \ pjsip.conf pjsip_notify.conf pjsip_wizard.conf prometheus.conf \ d261 6 @ 1.3 log @*: recursive bump for icu 78.1 @ text @d1 1 a1 1 # $NetBSD: Makefile,v 1.2 2025/12/01 04:26:02 jnemeth Exp $ d9 1 a9 1 PKGREVISION= 1 @ 1.2 log @Update to Asterisk 23.1.0. ## Change Log for Release asterisk-23.1.0 ### Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0) ### Summary: - Commits: 53 - Commit Authors: 17 - Issues Resolved: 37 - Security Advisories Resolved: 0 ### User Notes: - #### res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function. The STIR_SHAKEN_ATTESTATION dialplan function has been added which will allow suppressing attestation on a call-by-call basis regardless of the profile attached to the outgoing endpoint. - #### func_channel: Allow R/W of ADSI CPE capability setting. CHANNEL(adsicpe) can now be read or written to change the channels' ADSI CPE capability setting. - #### func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE() Added a new option to HANGUPCAUSE to access additional information about hangup reason. Reason headers from pjsip could be read using 'tech_extended' cause type. - #### func_math: Add DIGIT_SUM function. The DIGIT_SUM function can be used to return the digit sum of a number. - #### app_sf: Add post-digit timer option to ReceiveSF. The 't' option for ReceiveSF now allows for a timer since the last digit received, in addition to the number-wide timeout. - #### app_dial: Allow fractional seconds for dial timeouts. The answer and progress dial timeouts now have millisecond precision, instead of having to be whole numbers. - #### chan_dahdi: Add DAHDI_CHANNEL function. The DAHDI_CHANNEL function allows for getting/setting certain properties about DAHDI channels from the dialplan. ### Upgrade Notes: - #### app_queue.c: Fix error in Queue parameter documentation. As part of Asterisk 21, macros were removed from Asterisk. This resulted in argument order changing for the Queue dialplan application since the macro argument was removed. Upgrade notice was missed when this was done, so this upgrade note has been added to provide a record of such and a notice to users who may have not upgraded yet. - #### res_audiosocket: add message types for all slin sample rates New audiosocket message types 0x11 - 0x18 has been added for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and slin192 audio. External applications using audiosocket may need to be updated to support these message types if the audiosocket channel is created with one of these audio formats. - #### taskpool: Add taskpool API, switch Stasis to using it. The threadpool_* options in stasis.conf have now been deprecated though they continue to be read and used. They have been replaced with taskpool options that give greater control over the underlying taskpool used for stasis. ### Developer Notes: - #### chan_pjsip: Add technology-specific off-nominal hangup cause to events. A "tech_cause" parameter has been added to the ChannelHangupRequest and ChannelDestroyed ARI event messages and a "TechCause" parameter has been added to the HangupRequest, SoftHangupRequest and Hangup AMI event messages. For chan_pjsip, these will be set to the last SIP response status code for off-nominally terminated calls. The parameter is suppressed for nominal termination. - #### ARI: The bridges play and record APIs now handle sample rates > 8K correctly. The ARI /bridges/play and /bridges/record REST APIs have new parameters that allow the caller to specify the format to be used on the "Announcer" and "Recorder" channels respecitvely. - #### taskpool: Add taskpool API, switch Stasis to using it. The taskpool API has been added for common usage of a pool of taskprocessors. It is suggested to use this API instead of the threadpool+taskprocessor approach. ## Issue and Commit Detail: ### Closed Issues: - 781: [improvement]: Allow call by call disabling Stir/Shaken header inclusion - 1340: [bug]: comfort noise packet corrupted - 1419: [bug]: static code analysis issues in app_adsiprog.c - 1422: [bug]: static code analysis issues in apps/app_externalivr.c - 1425: [bug]: static code analysis issues in apps/app_queue.c - 1434: [improvement]: pbx_variables: Create real channel for dialplan eval CLI command - 1436: [improvement]: res_cliexec: Avoid unnecessary cast to char* - 1451: [bug]: ast_config_text_file_save2(): incorrect handling of deep/wide template inheritance - 1455: [new-feature]: chan_dahdi: Add DAHDI_CHANNEL function - 1467: [bug]: Crash in res_pjsip_refer during REFER progress teardown with PJSIP_TRANSFER_HANDLING(ari-only) - 1478: [improvement]: Stasis threadpool -> taskpool - 1479: [bug]: The ARI bridge play and record APIs limit audio bandwidth by forcing the slin8 format. - 1483: [improvement]: sig_analog: Eliminate possible timeout for Last Number Redial - 1485: [improvement]: func_scramble: Add example to XML documentation. - 1487: [improvement]: app_dial: Allow partial seconds to be used for dial timeouts - 1489: [improvement]: config_options.c: Improve misleading error message - 1491: [bug]: Segfault: `channelstorage_cpp` fast lookup without lock (`get_by_name_exact`/`get_by_uniqueid`) leads to UAF during hangup - 1493: [new-feature]: app_sf: Add post-digit timer option - 1496: [improvement]: dsp.c: Minor fixes to debug log messages - 1499: [new-feature]: func_math: Add function to return the digit sum - 1501: [improvement]: codec_builtin: Fix some inaccurate quality weights. - 1505: [improvement]: res_fax: Add XML documentation for channel variables - 1507: [improvement]: res_tonedetect: Minor formatting issue in documentation - 1509: [improvement]: res_fax.c — log debug error as debug, not regular log - 1510: [new-feature]: sig_analog: Allow '#' to end the inter-digit timeout when dialing. - 1514: [improvement]: func_channel: Allow R/W of ADSI CPE capability setting. - 1517: [improvement]: core_unreal: Preserve ADSI capability when dialing Local channels - 1519: [improvement]: app_dial / func_callerid: DNIS information is not propagated by Dial - 1525: [bug]: chan_websocket: fix use of raw payload variable for string comparison in process_text_message - 1534: [bug]: app_queue when using gosub breaks dialplan when going from 20 to 21, What's new in 21 doesn't mention it's a breaking change, - 1535: [bug]: chan_pjsip changes SSRC on WebRTC channels, which is unsupported by some browsers - 1536: [bug]: asterisk -rx connects to console instead of executing a command - 1539: [bug]: safe_asterisk without TTY doesn't log to file - 1544: [improvement]: While Receiving the MediaConnect Message Using External Media Over websocket ChannelID is Details are missing - 1554: [bug]: safe_asterisk recurses into subdirectories of startup.d after f97361 - 1559: [improvement]: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions. - 1578: [bug]: Deadlock with externalMedia custom channel id and cpp map channel backend @ text @d1 1 a1 1 # $NetBSD: Makefile,v 1.1 2025/10/27 06:58:36 jnemeth Exp $ d9 1 @ 1.1 log @comms/asterisk23: import asterisk-23.0.0 Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway). This is a standard version. It is scheduled to go to security fixes only on October 15th, 2026, and EOL on October 15th, 2027. See here for more information about Asterisk versions: https://docs.asterisk.org/About-the-Project/Asterisk-Versions/ @ text @d1 1 a1 1 # $NetBSD: Makefile,v 1.15 2025/10/27 04:35:13 jnemeth Exp $ d8 1 a8 1 DISTNAME= asterisk-23.0.0 d261 2 @