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locks; strict;
comment	@# @;


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desc
@@


1.179
log
@*: Recursive revbump from security/nettle-4.0
@
text
@# $NetBSD: Makefile,v 1.178 2026/04/10 08:41:34 wiz 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
#       and libjwt

DISTNAME=	asterisk-18.26.4
PKGREVISION=	4
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 perl:run pkg-config tar bash:run gsed
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
CONFIGURE_ARGS+=	--with-libjwt-bundled=yes
# XXX remove when lang/lua gets builtin.mk
CONFIGURE_ARGS+=	--without-lua
CONFIGURE_ENV+=		ac_cv_path_XMLSTARLET=":"

.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

.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+=	mgcp
# NOSIGPIPE is a temp variable, since PLIST.mgcp MUST remain undefined
# if the grep fails
.for dir in ${COMPILER_INCLUDE_DIRS}
.  if exists(${dir}/sys/socket.h)
NOSIGPIPE!=	${GREP} SO_NOSIGPIPE ${dir}/sys/socket.h || echo ""
.    if ${NOSIGPIPE} != ""
PLIST.mgcp=	yes
.    endif
.  endif
.endfor

.if ${OPSYS} == "SunOS"
PLIST.mgcp=	yes
.endif

PLIST_VARS+=		timerfd
.if defined(OPSYS_HAS_TIMERFD)
PLIST.timerfd=		yes
.else
CONFIGURE_ARGS+=	--without-timerfd
.endif

PLIST_VARS+=		unbound
# unbound 1.5 or later is required.`
.if ${OPSYS} == "NetBSD" && ${OPSYS_VERSION} > 090000 && \
    exists(/usr/include/unbound.h)
PLIST.unbound=		yes
.else
CONFIGURE_ARGS+=	--without-unbound
.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=	2.15.1
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}

# libjwt
LIBJWT_VERSION=	1.15.3
SITES.libjwt-${LIBJWT_VERSION}.tar.gz= \
	-https://raw.githubusercontent.com/asterisk/third-party/master/libjwt/${LIBJWT_VERSION}/libjwt-${LIBJWT_VERSION}.tar.gz
SITES.libjwt-${LIBJWT_VERSION}.md5= \
	-https://raw.githubusercontent.com/asterisk/third-party/master/libjwt/${LIBJWT_VERSION}/MD5SUM.TXT
DISTFILES+=	libjwt-${LIBJWT_VERSION}.tar.gz libjwt-${LIBJWT_VERSION}.md5

# 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/osp.conf.sample
SUBST_FILES.configs+=	configs/samples/phoneprov.conf.sample
SUBST_FILES.configs+=	configs/samples/res_config_sqlite.conf.sample
#SUBST_FILES.configs+=	configs/samples/sla.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|"

## XXX gross hack, gethostbyname_r on NETBSD is for internal use only
#SUBST_CLASSES.NetBSD+=		gethostbyname_r
#SUBST_STAGE.gethostbyname_r=	post-configure
#SUBST_FILES.gethostbyname_r=	include/asterisk/autoconfig.h
#SUBST_SED.gethostbyname_r=	-e "s|^\#define HAVE_GETHOSTBYNAME_R_5 1|\#undef HAVE_GETHOSTBYNAME_R_5|"
#SUBST_SED.gethostbyname_r+=	-e "s|^\#define HAVE_GETHOSTBYNAME_R_6 1|\#undef HAVE_GETHOSTBYNAME_R_6|"

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	\
	alsa.conf amd.conf app_mysql.conf app_skel.conf ari.conf	\
	ast_debug_tools.conf asterisk.adsi calendar.conf ccss.conf	\
	cdr.conf cdr_adaptive_odbc.conf cdr_custom.conf			\
	cdr_manager.conf cdr_mysql.conf cdr_odbc.conf cdr_pgsql.conf	\
	cdr_sqlite3_custom.conf cdr_syslog.conf cdr_tds.conf cel.conf	\
	cel_custom.conf cel_odbc.conf cel_pgsql.conf			\
	cel_sqlite3_custom.conf cel_tds.conf chan_dahdi.conf		\
	chan_mobile.conf cli.conf cli_aliases.conf cli_permissions.conf	\
	codecs.conf confbridge.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 mgcp.conf	\
	minivm.conf misdn.conf modules.conf motif.conf musiconhold.conf	\
	muted.conf ooh323.conf osp.conf oss.conf phone.conf		\
	phoneprov.conf pjproject.conf pjsip.conf pjsip_notify.conf	\
	pjsip_wizard.conf queuerules.conf queues.conf			\
	res_config_mysql.conf res_config_odbc.conf			\
	res_config_sqlite.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_pktccops.conf res_snmp.conf	\
	res_stun_monitor.conf resolver_unbound.conf rtp.conf say.conf	\
	sip.conf sip_notify.conf skinny.conf sla.conf smdi.conf		\
	sorcery.conf ss7.timers stasis.conf statsd.conf			\
	stir_shaken.conf telcordia-1.adsi udptl.conf unistim.conf	\
	users.conf voicemail.conf vpb.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 <sys/poll.h>" > ${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}/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}/ChangeLogs/ChangeLog-18.18.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.18.1.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.19.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.20.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.20.1.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.20.2.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.21.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.22.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.23.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.23.1.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.24.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.24.1.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.24.2.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.24.3.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.25.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.26.0.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.26.1.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.26.2.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.26.2.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.26.3.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.26.3.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.26.4.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/ChangeLog-18.26.4.html ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/historical/CHANGES ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/ChangeLogs/historical/ChangeLog ${DESTDIR}${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/SDL/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"
@


1.178
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 1.177 2026/02/06 10:04:20 wiz Exp $
d10 1
a10 1
PKGREVISION=	3
@


1.177
log
@*: recursive bump for nettle 4.0 shlib major bump
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.176 2026/01/07 08:46:04 wiz Exp $
d22 1
a22 1
OWNER=		jnemeth@@NetBSD.org
@


1.176
log
@*: recursive bump for icu 78.1
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.175 2025/10/05 19:25:44 js Exp $
d10 1
a10 1
PKGREVISION=	2
@


1.175
log
@*: rev bump for curl
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.174 2025/09/29 03:12:19 jnemeth Exp $
d10 1
a10 1
PKGREVISION=	1
@


1.174
log
@asterisk18:  Update to Asterisk 18.26.4.

pkgsrc change:
- fix PR pkg/59478

## Change Log for Release asterisk-18.26.4

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.4.html)

### Summary:

- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
  - [GHSA-557q-795j-wfx2](https://github.com/asterisk/asterisk/security/advisories/GHSA-557q-795j-wfx2): Resource exhaustion (DoS) vulnerability: remotely exploitable leak of RTP UDP ports and internal resources

### Commit Authors:

- George Joseph: (1)

## Issue and Commit Detail:

### Closed Issues:

  - !GHSA-557q-795j-wfx2: Resource exhaustion (DoS) vulnerability: remotely exploitable leak of RTP UDP ports and internal resources

### Commits By Author:

- #### George Joseph (1):
  - pjproject: Update bundled to 2.15.1.


### Commit List:

-  pjproject: Update bundled to 2.15.1.

### Commit Details:

#### pjproject: Update bundled to 2.15.1.
  Author: George Joseph
  Date:   2025-08-25

  This resolves a security issue where RTP ports weren't being released
  causing possible resource exhaustion issues.

  Resolves: #GHSA-557q-795j-wfx2
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.173 2025/08/30 22:44:44 wiz Exp $
d10 1
@


1.173
log
@*: recursive bump for tiff growing lerc dependency
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.172 2025/08/04 20:17:17 jnemeth Exp $
d9 1
a9 2
DISTNAME=	asterisk-18.26.3
PKGREVISION=	1
d53 1
d165 1
a165 1
PJPROJ_VERSION=	2.14.1
d323 2
@


1.172
log
@Update to Asterisk 18.26.3.  This is a security update.

## Change Log for Release asterisk-18.26.3

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.3.html)

### Summary:

- Commits: 2
- Commit Authors: 2
- Issues Resolved: 0
- Security Advisories Resolved: 2
  - [GHSA-mrq5-74j5-f5cr](https://github.com/asterisk/asterisk/security/advisories/GHSA-mrq5-74j5-f5cr): Remote DoS and possible RCE in asterisk/res/res_stir_shaken/verification.c
  - [GHSA-v9q8-9j8m-5xwp](https://github.com/asterisk/asterisk/security/advisories/GHSA-v9q8-9j8m-5xwp): Uncontrolled Search-Path Element in safe_asterisk script may allow local privilege escalation.

### User Notes:


### Upgrade Notes:

- #### safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
  The safe_asterisk script now checks that, if it was run by the
  root user, the /etc/asterisk/startup.d directory and all the files it contains
  are owned by root.  If the checks fail, safe_asterisk will exit with an error
  and Asterisk will not be started.  Additionally, the default logging
  destination is now stderr instead of tty "9" which probably won't exist
  in modern systems.


### Developer Notes:


### Commit Authors:

- George Joseph: (1)
- ThatTotallyRealMyth: (1)

## Issue and Commit Detail:

### Closed Issues:

  - !GHSA-mrq5-74j5-f5cr: Remote DoS and possible RCE in asterisk/res/res_stir_shaken/verification.c
  - !GHSA-v9q8-9j8m-5xwp: Uncontrolled Search-Path Element in safe_asterisk script may allow local privilege escalation.

### Commits By Author:

- #### George Joseph (1):
  - res_stir_shaken: Test for missing semicolon in Identity header.

- #### ThatTotallyRealMyth (1):
  - safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.


### Commit List:

-  safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
-  res_stir_shaken: Test for missing semicolon in Identity header.

### Commit Details:

#### safe_asterisk: Add ownership checks for /etc/asterisk/startup.d and its files.
  Author: ThatTotallyRealMyth
  Date:   2025-06-10

  UpgradeNote: The safe_asterisk script now checks that, if it was run by the
  root user, the /etc/asterisk/startup.d directory and all the files it contains
  are owned by root.  If the checks fail, safe_asterisk will exit with an error
  and Asterisk will not be started.  Additionally, the default logging
  destination is now stderr instead of tty "9" which probably won't exist
  in modern systems.

  Resolves: #GHSA-v9q8-9j8m-5xwp

#### res_stir_shaken: Test for missing semicolon in Identity header.
  Author: George Joseph
  Date:   2025-07-31

  ast_stir_shaken_vs_verify() now makes sure there's a semicolon in
  the Identity header to prevent a possible segfault.

  Resolves: #GHSA-mrq5-74j5-f5cr
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.171 2025/06/02 03:16:07 jnemeth Exp $
d10 1
@


1.171
log
@Update to Asterisk 18.26.2:

pkgsrc changes
- add gsed to USE_TOOLS
- appease pkglint somewhat


## Change Log for Release asterisk-18.26.2

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)

### Summary:

- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
  - [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw): Using malformed From header can forge identity with ";" or NULL in name portion
  - [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2): cli_permissions.conf: deny option does not work for disallowing shell commands

### User Notes:

- #### asterisk.c: Add option to restrict shell access from remote consoles.
  A new asterisk.conf option 'disable_remote_console_shell' has
  been added that, when set, will prevent remote consoles from executing
  shell commands using the '!' prefix.
  Resolves: #GHSA-c7p6-7mvq-8jq2


## Issue and Commit Detail:

### Closed Issues:

  - !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";" or NULL in name portion
  - !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for disallowing shell commands
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.170 2025/05/19 04:13:38 jnemeth Exp $
d9 1
a9 1
DISTNAME=	asterisk-18.26.2
d320 2
@


1.170
log
@Update to Asterisk 18.26.1


## Change Log for Release asterisk-18.26.1

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)

### Summary:

- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh): Path traversal via AMI ListCategories allows access to outside files

### User Notes:

- #### manager.c: Restrict ListCategories to the configuration directory.
  The ListCategories AMI action now restricts files to the
  configured configuration directory.


### Commit List:

-  manager.c: Restrict ListCategories to the configuration directory.

### Commit Details:

#### manager.c: Restrict ListCategories to the configuration directory.
  Author: Ben Ford
  Date:   2024-12-17

  When using the ListCategories AMI action, it was possible to traverse
  upwards through the directories to files outside of the configured
  configuration directory. This action is now restricted to the configured
  directory and an error will now be returned if the specified file is
  outside of this limitation.

  Resolves: #GHSA-33x6-fj46-6rfh

  UserNote: The ListCategories AMI action now restricts files to the
  configured configuration directory.


## Change Log for Release asterisk-18.26.0

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)

### Summary:

- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0

### User Notes:

- #### manager.c: Restrict ModuleLoad to the configured modules directory.
  The ModuleLoad AMI action now restricts modules to the
  configured modules directory.

- #### manager: Enhance event filtering for performance
  You can now perform more granular filtering on events
  in manager.conf using expressions like
  `eventfilter(name(Newchannel),header(Channel),method(starts_with)) = PJSIP/`
  This is much more efficient than
  `eventfilter = Event: Newchannel.*Channel: PJSIP/`
  Full syntax guide is in configs/samples/manager.conf.sample.

- #### db.c: Remove limit on family/key length
  The `ast_db_*()` APIs have had the 253 byte limit on
  "/family/key" removed and will now accept families and keys with a
  total length of up to SQLITE_MAX_LENGTH (currently 1e9!).  This
  affects the `DB*` dialplan applications, dialplan functions,
  manager actions and `databse` CLI commands.  Since the
  media_cache also uses the `ast_db_*()` APIs, you can now store
  resources with URIs longer than 253 bytes.


### Upgrade Notes:


## Issue and Commit Detail:

### Closed Issues:

  - 487: [bug]: Segfault possibly in ast_rtp_stop
  - 821: [bug]: app_dial:  The progress timeout doesn't cause Dial to exit
  - 881: [bug]: Long URLs are being rejected by the media cache because of an astdb key length limit
  - 882: [bug]: Value CHANNEL(userfield) is lost by BRIDGE_ENTER
  - 897: [improvement]: Restrict ModuleLoad AMI action to the modules directory
  - 900: [bug]: astfd.c: NULL pointer passed to fclose with nonnull attribute causes compilation failure
  - 902: [bug]: app_voicemail: Pager emails are ill-formatted when custom subject is used
  - 916: [bug]: Compilation errors on FreeBSD
  - 924: [bug]: dnsmgr.c: dnsmgr_refresh() should not flag change if IP address order changes
  - 928: [bug]: chan_dahdi: MWI while off-hook when hung up on after recall ring
  - 937: [bug]: Wrong format for sample config file 'geolocation.conf.sample'
  - 938: [bug]: memory leak - CBAnn leaks a small amount format_cap related memory for every confbridge


### Commit List:

-  app_dial: Fix progress timeout calculation with no answer timeout.
-  pjproject_bundled:  Tweaks to support out-of-tree development
-  chan_sip.c: Fix __sip_reliable_xmit build error
-  core_unreal.c: Fix memory leak in ast_unreal_new_channels()
-  dnsmgr.c: dnsmgr_refresh() incorrectly flags change with DNS round-robin
-  geolocation.sample.conf: Fix comment marker at end of file
-  func_base64.c: Ensure we set aside enough room for base64 encoded data.
-  app_dial: Fix progress timeout.
-  chan_dahdi: Never send MWI while off-hook.
-  manager.c: Add unit test for Originate app and appdata permissions
-  alembic: Drop redundant voicemail_messages index.
-  res_agi.c: Ensure SIGCHLD handler functions are properly balanced.
-  main, res, tests: Fix compilation errors on FreeBSD.
-  res_rtp_asterisk: Fix dtls timer issues causing FRACKs and SEGVs
-  manager.c: Restrict ModuleLoad to the configured modules directory.
-  res_agi.c: Prevent possible double free during `SPEECH RECOGNIZE`
-  cdr_custom: Allow absolute filenames.
-  astfd.c: Avoid calling fclose with NULL argument.
-  channel: Preserve CHANNEL(userfield) on masquerade.
-  cel_custom: Allow absolute filenames.
-  app_voicemail: Fix ill-formatted pager emails with custom subject.
-  res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery
-  Fix application references to Background
-  manager.conf.sample: Fix mathcing typo
-  manager: Enhance event filtering for performance
-  manager.c: Split XML documentation to manager_doc.xml
-  db.c: Remove limit on family/key length
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.169 2025/04/24 14:13:23 wiz Exp $
d9 1
a9 1
DISTNAME=	asterisk-18.26.1
d33 1
a33 1
USE_TOOLS+=		bison gmake perl:run pkg-config tar bash:run
d117 1
a117 1
.if !empty(MACHINE_PLATFORM:MSunOS-*-i386)
d130 1
a130 1
.  if (exists(${dir}/sys/socket.h))
d149 1
a149 1
PLIST_VARS+=	unbound
d153 1
a153 1
PLIST.unbound=	yes
d283 1
a283 1
.if (${OPSYS} == "Darwin" && exists(/usr/include/sys/poll.h))
d318 2
@


1.169
log
@*: recursive bump for jpeg -> libjpeg-turbo switch
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.168 2025/04/19 07:57:33 wiz Exp $
d9 1
a9 2
DISTNAME=	asterisk-18.25.0
PKGREVISION=	7
d316 2
@


1.168
log
@*: recursive bump for default Kerberos implementation switch
@
text
@d1 1
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# $NetBSD: Makefile,v 1.167 2025/04/17 21:49:57 wiz Exp $
d10 1
a10 1
PKGREVISION=	6
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@Upgrade to Asterisk 18.25.0.

## Change Log for Release asterisk-18.25.0

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0)
 - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz)
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

### Summary:

- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0

### User Notes:

- #### res_pjsip_notify: add dialplan application
  A new dialplan application PJSIPNotify is now available
  which can send SIP NOTIFY requests from the dialplan.
  The pjsip send notify CLI command has also been enhanced to allow
  sending NOTIFY messages to a specific channel. Syntax:
  pjsip send notify <option> channel <channel>

- #### channel: Add multi-tenant identifier.
  tenantid has been added to channels. It can be read in
  dialplan via CHANNEL(tenantid), and it can be set using
  Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to
  use the new tenantid option for pjsip endpoints (e.g., tenantid=My
  tenant ID) so that it will show up in Newchannel events. You can set it
  like any other channel variable using set_var in pjsip.conf as well, but
  note that this will NOT show up in Newchannel events. Tenant ID is also
  available in CDR and can be accessed with CDR(tenantid). The peer tenant
  ID can also be accessed with CDR(peertenantid). CEL includes tenant ID
  as well if it has been set.

- #### res_pjsip_config_wizard.c: Refactor load process
  The res_pjsip_config_wizard.so module can now be reloaded.


### Upgrade Notes:

- #### channel: Add multi-tenant identifier.
  A new versioned struct (ast_channel_initializers) has been
  added that gets passed to __ast_channel_alloc_ap. The new function
  ast_channel_alloc_with_initializers should be used when creating
  channels that require the use of this struct. Currently the only value
  in the struct is for tenantid, but now more fields can be added to the
  struct as necessary rather than the __ast_channel_alloc_ap function. A
  new option (tenantid) has been added to endpoints in pjsip.conf as well.
  CEL has had its version bumped to include tenant ID.


### Commit Authors:

- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)

## Issue and Commit Detail:

### Closed Issues:

  - 740: [new-feature]: Add multi-tenant identifier to chan_pjsip
  - 763: [bug]: autoservice thread stuck in an endless sleep
  - 780: [bug]: Infinite loop of "Indicated Video Update", max CPU usage
  - 799: [improvement]: Add PJSIPNOTIFY dialplan application
  - 801: [bug]: res_stasis: Occasional 200ms delay adding channel to a bridge
  - 809: [bug]: CLI stir_shaken show verification kills asterisk
  - 816: [bug]: res_pjsip_config_wizard doesn't load properly if res_pjsip is loaded first
  - 831: [bug]: app_voicemail ODBC
  - 845: [bug]: Buffer overflow in handling of security mechanisms in res_pjsip
  - 847: [bug]: Asterisk not using negotiated fall-back 8K digits
  - 854: [bug]:  wrong properties in stir_shaken.conf.sample
  - 856: [bug]: res_pjsip_sdp_rtp leaks astobj2 ast_format
  - 861: [bug]: ChanSpy unable to read audiohook read direction frame when no packet lost on both side of the call
  - 876: [bug]: ChanSpy unable to write whisper_audiohook when set flag OPTION_READONLY
  - 879: [bug]: res_stir_shaken/verification.c: Getting verification errors when global_disable=yes
  - 884: [bug]: A ':' at the top of in stir_shaken.conf make Asterisk producing a core file when starting
  - 889: [bug]: res_stir_shaken/verification.c has a stale include for jansson.h that can cause compilation to fail
  - 904: [bug]: stir_shaken: attest_level isn't being propagated correctly from attestation to profile to tn

### Commits By Author:

- #### Alexei Gradinari (2):
  - res_pjsip_sdp_rtp fix leaking astobj2 ast_format
  - autoservice: Do not sleep if autoservice_stop is called within autoservice thr..

- #### Ben Ford (1):
  - channel: Add multi-tenant identifier.

- #### Cade Parker (1):
  - chan_mobile: decrease CHANNEL_FRAME_SIZE to prevent delay

- #### George Joseph (10):
  - bridge_softmix: Fix queueing VIDUPDATE control frames
  - res_pjsip_config_wizard.c: Refactor load process
  - stir_shaken: CRL fixes and a new CLI command
  - manager.c: Fix FRACK when doing CoreShowChannelMap in DEVMODE
  - stir_shaken.conf.sample: Fix bad references to private_key_path
  - security_agreements.c: Refactor the to_str functions and fix a few other bugs
  - res_stir_shaken: Check for disabled before param validation
  - res_stir_shaken.c: Fix crash when stir_shaken.conf is invalid
  - res_stir_shaken: Remove stale include for jansson.h in verification.c
  - stir_shaken: Fix propagation of attest_level and a few other values

- #### Jaco Kroon (1):
  - configure:  Use . file rather than source file.

- #### Jean-Denis Girard (1):
  - app_voicemail: Fix sql insert mismatch caused by cherry-pick

- #### Mike Bradeen (3):
  - res_stasis: fix intermittent delays on adding channel to bridge
  - res_pjsip_notify: add dialplan application
  - res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch

- #### Sean Bright (2):
  - alembic: Make 'revises' header comment match reality.
  - res_pjsip_logger.c: Fix 'OPTIONS' tab completion.

- #### Tinet-mucw (3):
  - res_pjsip_sdp_rtp.c: Fix DTMF Handling in Re-INVITE with dtmf_mode set to auto
  - app_chanspy.c: resolving the issue with audiohook direction read
  - app_chanspy.c: resolving the issue writing frame to whisper audiohook.


### Commit List:

-  stir_shaken: Fix propagation of attest_level and a few other values
-  res_stir_shaken: Remove stale include for jansson.h in verification.c
-  res_stir_shaken.c: Fix crash when stir_shaken.conf is invalid
-  res_stir_shaken: Check for disabled before param validation
-  app_chanspy.c: resolving the issue writing frame to whisper audiohook.
-  app_voicemail: Fix sql insert mismatch caused by cherry-pick
-  res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch
-  app_chanspy.c: resolving the issue with audiohook direction read
-  security_agreements.c: Refactor the to_str functions and fix a few other bugs
-  res_pjsip_sdp_rtp fix leaking astobj2 ast_format
-  stir_shaken.conf.sample: Fix bad references to private_key_path
-  res_pjsip_logger.c: Fix 'OPTIONS' tab completion.
-  alembic: Make 'revises' header comment match reality.
-  chan_mobile: decrease CHANNEL_FRAME_SIZE to prevent delay
-  res_pjsip_notify: add dialplan application
-  manager.c: Fix FRACK when doing CoreShowChannelMap in DEVMODE
-  channel: Add multi-tenant identifier.
-  configure:  Use . file rather than source file.
-  res_stasis: fix intermittent delays on adding channel to bridge
-  res_pjsip_sdp_rtp.c: Fix DTMF Handling in Re-INVITE with dtmf_mode set to auto
-  stir_shaken: CRL fixes and a new CLI command
-  res_pjsip_config_wizard.c: Refactor load process
-  bridge_softmix: Fix queueing VIDUPDATE control frames

### Commit Details:

#### stir_shaken: Fix propagation of attest_level and a few other values
  Author: George Joseph
  Date:   2024-09-24

  attest_level, send_mky and check_tn_cert_public_url weren't
  propagating correctly from the attestation object to the profile
  and tn.

  * In the case of attest_level, the enum needed to be changed
  so the "0" value (the default) was "NOT_SET" instead of "A".  This
  now allows the merging of the attestation object, profile and tn
  to detect when a value isn't set and use the higher level value.

  * For send_mky and check_tn_cert_public_url, the tn default was
  forced to "NO" which always overrode the profile and attestation
  objects.  Their defaults are now "NOT_SET" so the propagation
  happens correctly.

  * Just to remove some redundant code in tn_config.c, a bunch of calls to
  generate_sorcery_enum_from_str() and generate_sorcery_enum_to_str() were
  replaced with a single call to generate_acfg_common_sorcery_handlers().

  Resolves: #904

#### res_stir_shaken: Remove stale include for jansson.h in verification.c
  Author: George Joseph
  Date:   2024-09-17

  verification.c had an include for jansson.h left over from previous
  versions of the module.  Since res_stir_shaken no longer has a
  dependency on jansson, the bundled version wasn't added to GCC's
  include path so if you didn't also have a jansson development package
  installed, the compile would fail.  Removing the stale include
  was the only thing needed.

  Resolves: #889

#### res_stir_shaken.c: Fix crash when stir_shaken.conf is invalid
  Author: George Joseph
  Date:   2024-09-13

  * If the call to ast_config_load() returns CONFIG_STATUS_FILEINVALID,
  check_for_old_config() now returns LOAD_DECLINE instead of continuing
  on with a bad pointer.

  * If CONFIG_STATUS_FILEMISSING is returned, check_for_old_config()
  assumes the config is being loaded from realtime and now returns
  LOAD_SUCCESS.  If it's actually not being loaded from realtime,
  sorcery will catch that later on.

  * Also refactored the error handling in load_module() a bit.

  Resolves: #884

#### res_stir_shaken: Check for disabled before param validation
  Author: George Joseph
  Date:   2024-09-11

  For both attestation and verification, we now check whether they've
  been disabled either globally or by the profile before validating
  things like callerid, orig_tn, dest_tn, etc.  This prevents useless
  error messages.

  Resolves: #879

#### app_chanspy.c: resolving the issue writing frame to whisper audiohook.
  Author: Tinet-mucw
  Date:   2024-09-10

  ChanSpy(${channel}, qEoSw): because flags set o, ast_audiohook_set_frame_feed_direction(audiohook, AST_AUDIOHOOK_DIRECTION_READ); this will effect whisper audiohook and spy audiohook, this makes writing frame to whisper audiohook impossible. So add function start_whispering to starting whisper audiohook.

  Resolves: #876

#### autoservice: Do not sleep if autoservice_stop is called within autoservice thr..
  Author: Alexei Gradinari
  Date:   2024-09-04

  It's possible that ast_autoservice_stop is called within the autoservice thread.
  In this case the autoservice thread is stuck in an endless sleep.

  To avoid endless sleep ast_autoservice_stop must check that it's not called
  within the autoservice thread.

  Fixes: #763

#### app_voicemail: Fix sql insert mismatch caused by cherry-pick
  Author: Jean-Denis Girard
  Date:   2024-08-07

  When commit e8c9cb80 was cherry-picked in from master, the
  fact that the 20 and 18 branches still had the old "macrocontext"
  column wasn't taken into account so the number of named parameters
  didn't match the number of '?' placeholders.  They do now.

  We also now use ast_asprintf to create the full mailbox query SQL
  statement instead of trying to calculate the proper length ourselves.

  Resolves: #831

#### res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch
  Author: Mike Bradeen
  Date:   2024-08-21

  When Asterisk sends an offer to Bob that includes 48K and 8K codecs with
  matching 4733 offers, Bob may want to use the 48K audio codec but can not
  accept 48K digits and so negotiates for a mixed set.

  Asterisk will now check Bob's offer to make sure Bob has indicated this is
  acceptible and if not, will use Bob's preference.

  Fixes: #847

#### app_chanspy.c: resolving the issue with audiohook direction read
  Author: Tinet-mucw
  Date:   2024-08-30

  ChanSpy(${channel}, qEoS): When chanspy spy the direction read, reading frame is often failed when reading direction read audiohook. because chanspy only read audiohook direction read; write_factory_ms will greater than 100ms soon, then ast_slinfactory_flush will being called, then direction read will fail.

  Resolves: #861

#### security_agreements.c: Refactor the to_str functions and fix a few other bugs
  Author: George Joseph
  Date:   2024-08-17

  * A static array of security mechanism type names was created.

  * ast_sip_str_to_security_mechanism_type() was refactored to do
    a lookup in the new array instead of using fixed "if/else if"
    statments.

  * security_mechanism_to_str() and ast_sip_security_mechanisms_to_str()
    were refactored to use ast_str instead of a fixed length buffer
    to store the result.

  * ast_sip_security_mechanism_type_to_str was removed in favor of
    just referencing the new type name array.  Despite starting with
    "ast_sip_", it was a static function so removing it doesn't affect
    ABI.

  * Speaking of "ast_sip_", several other static functions that
    started with "ast_sip_" were renamed to avoid confusion about
    their public availability.

  * A few VECTOR free loops were replaced with AST_VECTOR_RESET().

  * Fixed a meomry leak in pjsip_configuration.c endpoint_destructor
    caused by not calling ast_sip_security_mechanisms_vector_destroy().

  * Fixed a memory leak in res_pjsip_outbound_registration.c
    add_security_headers() caused by not specifying OBJ_NODATA in
    an ao2_callback.

  * Fixed a few ao2_callback return code misuses.

  Resolves: #845

#### res_pjsip_sdp_rtp fix leaking astobj2 ast_format
  Author: Alexei Gradinari
  Date:   2024-08-23

  PR #700 added a preferred_format for the struct ast_rtp_codecs,
  but when set the preferred_format it leaks an astobj2 ast_format.
  In the next code
  ast_rtp_codecs_set_preferred_format(&codecs, ast_format_cap_get_format(joint, 0));
  both functions ast_rtp_codecs_set_preferred_format
  and ast_format_cap_get_format increases the ao2 reference count.

  Fixes: #856

#### stir_shaken.conf.sample: Fix bad references to private_key_path
  Author: George Joseph
  Date:   2024-08-22

  They should be private_key_file.

  Resolves: #854

#### res_pjsip_logger.c: Fix 'OPTIONS' tab completion.
  Author: Sean Bright
  Date:   2024-08-19

  Fixes #843


#### alembic: Make 'revises' header comment match reality.
  Author: Sean Bright
  Date:   2024-08-17


#### chan_mobile: decrease CHANNEL_FRAME_SIZE to prevent delay
  Author: Cade Parker
  Date:   2024-08-07

  On modern Bluetooth devices or lower-powered asterisk servers, decreasing the channel frame size significantly improves latency and delay on outbound calls with only a mild sacrifice to the quality of the call (the frame size before was massive overkill to begin with)


#### res_pjsip_notify: add dialplan application
  Author: Mike Bradeen
  Date:   2024-07-09

  Add dialplan application PJSIPNOTIFY to send either pre-configured
  NOTIFY messages from pjsip_notify.conf or with headers defined in
  dialplan.

  Also adds the ability to send pre-configured NOTIFY commands to a
  channel via the CLI.

  Resolves: #799

  UserNote: A new dialplan application PJSIPNotify is now available
  which can send SIP NOTIFY requests from the dialplan.

  The pjsip send notify CLI command has also been enhanced to allow
  sending NOTIFY messages to a specific channel. Syntax:

  pjsip send notify <option> channel <channel>


#### manager.c: Fix FRACK when doing CoreShowChannelMap in DEVMODE
  Author: George Joseph
  Date:   2024-08-08

  If you run an AMI CoreShowChannelMap on a channel that isn't in a
  bridge and you're in DEVMODE, you can get a FRACK because the
  bridge id is empty.  We now simply return an empty list for that
  request.


#### channel: Add multi-tenant identifier.
  Author: Ben Ford
  Date:   2024-05-21

  This patch introduces a new identifier for channels: tenantid. It's
  a stringfield on the channel that can be used for general purposes. It
  will be inherited by other channels the same way that linkedid is.

  You can set tenantid in a few ways. The first is to set it in the
  dialplan with the Set and CHANNEL functions:

  exten => example,1,Set(CHANNEL(tenantid)=My tenant ID)

  It can also be accessed via CHANNEL:

  exten => example,2,NoOp(CHANNEL(tenantid))

  Another method is to use the new tenantid option for pjsip endpoints in
  pjsip.conf:

  [my_endpoint]
  type=endpoint
  tenantid=My tenant ID

  This is considered the best approach since you will be able to see the
  tenant ID as early as the Newchannel event.

  It can also be set using set_var in pjsip.conf on the endpoint like
  setting other channel variable:

  set_var=CHANNEL(tenantid)=My tenant ID

  Note that set_var will not show tenant ID on the Newchannel event,
  however.

  Tenant ID has also been added to CDR. It's read-only and can be accessed
  via CDR(tenantid). You can also get the tenant ID of the last channel
  communicated with via CDR(peertenantid).

  Tenant ID will also show up in CEL records if it has been set, and the
  version number has been bumped accordingly.

  Fixes: #740

  UserNote: tenantid has been added to channels. It can be read in
  dialplan via CHANNEL(tenantid), and it can be set using
  Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to
  use the new tenantid option for pjsip endpoints (e.g., tenantid=My
  tenant ID) so that it will show up in Newchannel events. You can set it
  like any other channel variable using set_var in pjsip.conf as well, but
  note that this will NOT show up in Newchannel events. Tenant ID is also
  available in CDR and can be accessed with CDR(tenantid). The peer tenant
  ID can also be accessed with CDR(peertenantid). CEL includes tenant ID
  as well if it has been set.

  UpgradeNote: A new versioned struct (ast_channel_initializers) has been
  added that gets passed to __ast_channel_alloc_ap. The new function
  ast_channel_alloc_with_initializers should be used when creating
  channels that require the use of this struct. Currently the only value
  in the struct is for tenantid, but now more fields can be added to the
  struct as necessary rather than the __ast_channel_alloc_ap function. A
  new option (tenantid) has been added to endpoints in pjsip.conf as well.
  CEL has had its version bumped to include tenant ID.


#### configure:  Use . file rather than source file.
  Author: Jaco Kroon
  Date:   2024-08-05

  source is a bash concept, so when /bin/sh points to another shell the
  existing construct won't work.

  Reference: https://bugs.gentoo.org/927055
  Signed-off-by: Jaco Kroon <jaco@@uls.co.za>

#### res_stasis: fix intermittent delays on adding channel to bridge
  Author: Mike Bradeen
  Date:   2024-07-10

  Previously, on command execution, the control thread was awoken by
  sending a SIGURG. It was found that this still resulted in some
  instances where the thread was not immediately awoken.

  This change instead sends a null frame to awaken the control thread,
  which awakens the thread more consistently.

  Resolves: #801

#### res_pjsip_sdp_rtp.c: Fix DTMF Handling in Re-INVITE with dtmf_mode set to auto
  Author: Tinet-mucw
  Date:   2024-08-02

  When the endpoint dtmf_mode is set to auto, a SIP request is sent to the UAC, and the SIP SDP from the UAC does not include the telephone-event. Later, the UAC sends an INVITE, and the SIP SDP includes the telephone-event. In this case, DTMF should be sent by RFC2833 rather than using inband signaling.

  Resolves: asterisk#826

#### stir_shaken: CRL fixes and a new CLI command
  Author: George Joseph
  Date:   2024-07-19

  * Fixed a bug in crypto_show_cli_store that was causing asterisk
  to crash if there were certificate revocation lists in the
  verification certificate store.  We're also now prefixing
  certificates with "Cert:" and CRLs with "CRL:" to distinguish them
  in the list.

  * Added 'untrusted_cert_file' and 'untrusted_cert_path' options
  to both verification and profile objects.  If you have CRLs that
  are signed by a different CA than the incoming X5U certificate
  (indirect CRL), you'll need to provide the certificate of the
  CRL signer here.  Thse will show up as 'Untrusted" when showing
  the verification or profile objects.

  * Fixed loading of crl_path.  The OpenSSL API we were using to
  load CRLs won't actually load them from a directory, only a file.
  We now scan the directory ourselves and load the files one-by-one.

  * Fixed the verification flags being set on the certificate store.
    - Removed the CRL_CHECK_ALL flag as this was causing all certificates
      to be checked for CRL extensions and failing to verify the cert if
      there was none.  This basically caused all certs to fail when a CRL
      was provided via crl_file or crl_path.
    - Added the EXTENDED_CRL_SUPPORT flag as it is required to handle
      indirect CRLs.

  * Added a new CLI command...
  `stir_shaken verify certificate_file <certificate_file> [ <profile> ]`
  which will assist troubleshooting certificate problems by allowing
  the user to manually verify a certificate file against either the
  global verification certificate store or the store for a specific
  profile.

  * Updated the XML documentation and the sample config file.

  Resolves: #809

#### res_pjsip_config_wizard.c: Refactor load process
  Author: George Joseph
  Date:   2024-07-23

  The way we have been initializing the config wizard prevented it
  from registering its objects if res_pjsip happened to load
  before it.

  * We now use the object_type_registered sorcery observer to kick
  things off instead of the wizard_mapped observer.

  * The load_module function now checks if res_pjsip has been loaded
  already and if it was it fires the proper observers so the objects
  load correctly.

  Resolves: #816

  UserNote: The res_pjsip_config_wizard.so module can now be reloaded.

#### bridge_softmix: Fix queueing VIDUPDATE control frames
  Author: George Joseph
  Date:   2024-07-17

  softmix_bridge_write_control() now calls ast_bridge_queue_everyone_else()
  with the bridge_channel so the VIDUPDATE control frame isn't echoed back.

  softmix_bridge_write_control() was setting bridge_channel to NULL
  when calling ast_bridge_queue_everyone_else() for VIDUPDATE control
  frames.  This was causing the frame to be echoed back to the
  channel it came from.  In certain cases, like when two channels or
  bridges are being recorded, this can cause a ping-pong effect that
  floods the system with VIDUPDATE control frames.

  Resolves: #780


## Change Log for Release asterisk-18.24.3

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
 - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

### Summary:

- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9): A malformed Contact or Record-Route URI in an incoming SIP request can cause Asterisk to crash when res_resolver_unbound is used

### User Notes:


### Upgrade Notes:


### Commit Authors:

- George Joseph: (1)

## Issue and Commit Detail:

### Closed Issues:

  - !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming SIP request can cause Asterisk to crash when res_resolver_unbound is used

### Commits By Author:

- #### George Joseph (1):
  - res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback


### Commit List:

-  res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback

### Commit Details:

#### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
  Author: George Joseph
  Date:   2024-08-12

  The ub_result pointer passed to unbound_resolver_callback by
  libunbound can be NULL if the query was for something malformed
  like `.1` or `[.1]`.  If it is, we now set a 'ns_r_formerr' result
  and return instead of crashing with a SEGV.  This causes pjproject
  to simply cancel the transaction with a "No answer record in the DNS
  response" error.  The existing "off nominal" unit test was also
  updated to check this condition.

  Although not necessary for this fix, we also made
  ast_dns_resolver_completed() tolerant of a NULL result.

  Resolves: GHSA-v428-g3cw-7hv9


## Change Log for Release asterisk-18.24.2

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
 - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

### Summary:

- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
  - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44): Write=originate, is sufficient permissions for code execution / System() dialplan

### User Notes:


### Upgrade Notes:


### Commit Authors:

- George Joseph: (1)

## Issue and Commit Detail:

### Closed Issues:

  - !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code execution / System() dialplan

### Commits By Author:

- #### George Joseph (1):
  - manager.c: Add entries to Originate blacklist


### Commit List:

-  manager.c: Add entries to Originate blacklist

### Commit Details:

#### manager.c: Add entries to Originate blacklist
  Author: George Joseph
  Date:   2024-07-22

  Added Reload and DBdeltree to the list of dialplan application that
  can't be executed via the Originate manager action without also
  having write SYSTEM permissions.

  Added CURL, DB*, FILE, ODBC and REALTIME* to the list of dialplan
  functions that can't be executed via the Originate manager action
  without also having write SYSTEM permissions.

  If the Queue application is attempted to be run by the Originate
  manager action and an AGI parameter is specified in the app data,
  it'll be rejected unless the manager user has either the AGI or
  SYSTEM permissions.

  Resolves: #GHSA-c4cg-9275-6w44


## Change Log for Release asterisk-18.24.1

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1)
 - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz)
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

### Summary:

- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0

### User Notes:


### Upgrade Notes:


### Commit Authors:

- George Joseph: (2)

## Issue and Commit Detail:

### Closed Issues:

  - 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using "make samples"
  - 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -> 20.9.0

### Commits By Author:

- #### George Joseph (2):
  - voicemail.conf.sample: Fix ':' comment typo
  - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()


### Commit List:

-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo

### Commit Details:

#### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
  Author: George Joseph
  Date:   2024-07-25

  There can be empty slots in payload_mapping_tx corresponding to
  dynamic payload types that haven't been seen before so we now
  check for NULL before attempting to use 'type' in the call to
  ast_format_cmp.

  Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()

  Resolves: #822

#### voicemail.conf.sample: Fix ':' comment typo
  Author: George Joseph
  Date:   2024-07-24

  ...and removed an errant trailing space.

  Resolves: #819


## Change Log for Release asterisk-18.24.0

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
 - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)

### Summary:

- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0

### User Notes:

- #### app_voicemail_odbc: Allow audio to be kept on disk
  This commit adds a new voicemail.conf option
  'odbc_audio_on_disk' which when set causes the ODBC variant of
  app_voicemail_odbc to leave the message and greeting audio files
  on disk and only store the message metadata in the database.
  Much more information can be found in the voicemail.conf.sample
  file.

- #### app_queue:  Add option to not log Restricted Caller ID to queue_log
  Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID
  will be stored in the queue log.
  If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted.

- #### pbx.c: expand fields width of "core show hints"
  The fields width of "core show hints" were increased.
  The width of "extension" field to 30 characters and
  the width of the "device state id" field to 60 characters.

- #### rtp_engine: add support for multirate RFC2833 digits
  No change in configuration is required in order to enable this
  feature. Endpoints configured to use RFC2833 will automatically have this
  enabled. If the endpoint does not support this, it should not include it in
  the SDP offer/response.
  Resolves: #699


### Upgrade Notes:

- #### app_queue:  Add option to not log Restricted Caller ID to queue_log
  Add a new column to the queues table:
  queue_log_option_log_restricted ENUM('0','1','off','on','false','true','no','yes')
  to control whether the Restricted Caller ID will be stored in the queue log.


### Commit Authors:

- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)

## Issue and Commit Detail:

### Closed Issues:

  - 699: [improvement]: Add support for multi-rate DTMF
  - 736: [bug]: Seg fault on CLI after PostgreSQL CDR module fails to load for a second time
  - 765: [improvement]: Add option to not log Restricted Caller ID to queue_log
  - 770: [improvement]: pbx.c: expand fields width of "core show hints"
  - 776: [bug] DTMF broken after rtp_engine: add support for multirate RFC2833 digits commit
  - 783: [bug]: Under certain circumstances a channel snapshot can get orphaned in the cache
  - 789: [bug]: Mediasec headers aren't sent on outgoing INVITEs
  - 797: [bug]:

### Commits By Author:

- ### Alexei Gradinari (2):
  - pbx.c: expand fields width of "core show hints"
  - app_queue:  Add option to not log Restricted Caller ID to queue_log

- ### George Joseph (4):
  - app_voicemail_odbc: Allow audio to be kept on disk
  - stasis_channels: Use uniqueid and name to delete old snapshots
  - security_agreement.c: Always add the Require and Proxy-Require headers
  - ast-db-manage: Remove duplicate enum creation

- ### Igor Goncharovsky (1):
  - res_pjsip_path.c: Fix path when dialing using PJSIP_DIAL_CONTACTS()

- ### Mike Bradeen (2):
  - rtp_engine: add support for multirate RFC2833 digits
  - res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits

- ### Sean Bright (7):
  - file.h: Rename function argument to avoid C++ keyword clash.
  - bundled_pjproject: Disable UPnP support.
  - asterisk.c: Don't log an error if .asterisk_history does not exist.
  - xml.c: Update deprecated libxml2 API usage.
  - manager.c: Properly terminate `CoreShowChannelMap` event.
  - pjsip: Add PJSIP_PARSE_URI_FROM dialplan function.
  - logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds.

- ### Tinet-mucw (1):
  - bridge_basic.c: Make sure that ast_bridge_channel is not destroyed while itera..

- ### Walter Doekes (1):
  - chan_ooh323: Fix R/0 typo in docs

- ### chrsmj (1):
  - cdr_pgsql: Fix crash when the module fails to load multiple times.


### Commit List:

-  res_pjsip_path.c: Fix path when dialing using PJSIP_DIAL_CONTACTS()
-  res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits
-  ast-db-manage: Remove duplicate enum creation
-  security_agreement.c: Always add the Require and Proxy-Require headers
-  logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds.
-  stasis_channels: Use uniqueid and name to delete old snapshots
-  app_voicemail_odbc: Allow audio to be kept on disk
-  app_queue:  Add option to not log Restricted Caller ID to queue_log
-  pbx.c: expand fields width of "core show hints"
-  pjsip: Add PJSIP_PARSE_URI_FROM dialplan function.
-  manager.c: Properly terminate `CoreShowChannelMap` event.
-  xml.c: Update deprecated libxml2 API usage.
-  cdr_pgsql: Fix crash when the module fails to load multiple times.
-  asterisk.c: Don't log an error if .asterisk_history does not exist.
-  chan_ooh323: Fix R/0 typo in docs
-  bundled_pjproject: Disable UPnP support.
-  file.h: Rename function argument to avoid C++ keyword clash.
-  rtp_engine: add support for multirate RFC2833 digits

### Commit Details:

#### res_pjsip_path.c: Fix path when dialing using PJSIP_DIAL_CONTACTS()
  Author: Igor Goncharovsky
  Date:   2024-05-12

  When using the PJSIP_DIAL_CONTACTS() function for use in the Dial()
  command, the contacts are returned in text form, so the input to
  the path_outgoing_request() function is a contact value of NULL.
  The issue was reported in ASTERISK-28211, but was not actually fixed
  in ASTERISK-30100. This fix brings back the code that was previously
  removed and adds code to search for a contact to extract the path
  value from it.


#### res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits
  Author: Mike Bradeen
  Date:   2024-06-21

  After change made in 624f509 to add support for non 8K RFC 4733/2833 digits,
  Asterisk would only accept RFC 4733/2833 offers that matched the sample rate of
  the negotiated codec(s).

  This change allows Asterisk to accept 8K RFC 4733/2833 offers if the UAC
  offfers 8K RFC 4733/2833 but negotiates for a non 8K bitrate codec.

  A number of corresponding tests in tests/channels/pjsip/dtmf_sdp also needed to
  be re-written to allow for these scenarios.

  Fixes: #776

#### ast-db-manage: Remove duplicate enum creation
  Author: George Joseph
  Date:   2024-07-08

  Remove duplicate creation of ast_bool_values from
  2b7c507d7d12_add_queue_log_option_log_restricted_.py.  This was
  causing alembic upgrades to fail since the enum was already created
  in fe6592859b85_fix_mwi_subscribe_replaces_.py back in 2018.

  Resolves: #797

#### security_agreement.c: Always add the Require and Proxy-Require headers
  Author: George Joseph
  Date:   2024-07-03

  The `Require: mediasec` and `Proxy-Require: mediasec` headers need
  to be sent whenever we send `Security-Client` or `Security-Verify`
  headers but the logic to do that was only in add_security_headers()
  in res_pjsip_outbound_register.  So while we were sending them on
  REGISTER requests, we weren't sending them on INVITE requests.

  This commit moves the logic to send the two headers out of
  res_pjsip_outbound_register:add_security_headers() and into
  security_agreement:ast_sip_add_security_headers().  This way
  they're always sent when we send `Security-Client` or
  `Security-Verify`.

  Resolves: #789

#### logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds.
  Author: Sean Bright
  Date:   2024-06-29

  Fixes #785


#### stasis_channels: Use uniqueid and name to delete old snapshots
  Author: George Joseph
  Date:   2024-05-08

  Whenver a new channel snapshot is created or when a channel is
  destroyed, we need to delete any existing channel snapshot from
  the snapshot cache.  Historically, we used the channel->snapshot
  pointer to delete any existing snapshots but this has two issues.

  First, if something (possibly ast_channel_internal_swap_snapshots)
  sets channel->snapshot to NULL while there's still a snapshot in
  the cache, we wouldn't be able to delete it and it would be orphaned
  when the channel is destroyed.  Since we use the cache to list
  channels from the CLI, AMI and ARI, it would appear as though the
  channel was still there when it wasn't.

  Second, since there are actually two caches, one indexed by the
  channel's uniqueid, and another indexed by the channel's name,
  deleting from the caches by pointer requires a sequential search of
  all of the hash table buckets in BOTH caches to find the matching
  snapshots.  Not very efficient.

  So, we now delete from the caches using the channel's uniqueid
  and name.  This solves both issues.

  This doesn't address how channel->snapshot might have been set
  to NULL in the first place because although we have concrete
  evidence that it's happening, we haven't been able to reproduce it.

  Resolves: #783

#### app_voicemail_odbc: Allow audio to be kept on disk
  Author: George Joseph
  Date:   2024-04-09

  This commit adds a new voicemail.conf option 'odbc_audio_on_disk'
  which when set causes the ODBC variant of app_voicemail to leave
  the message and greeting audio files on disk and only store the
  message metadata in the database.  This option came from a concern
  that the database could grow to large and cause remote access
  and/or replication to become slow.  In a clustering situation
  with this option, all asterisk instances would share the same
  database for the metadata and either use a shared filesystem
  or other filesystem replication service much more suitable
  for synchronizing files.

  The changes to app_voicemail to implement this feature were actually
  quite small but due to the complexity of the module, the actual
  source code changes were greater.  They fall into the following
  categories:

  * Tracing.  The module is so complex that it was impossible to
  figure out the path taken for various scenarios without the addition
  of many SCOPE_ENTER, SCOPE_EXIT and ast_trace statements, even in
  code that's not related to the functional change.  Making this worse
  was the fact that many "if" statements in this module didn't use
  braces.  Since the tracing macros add multiple statements, many "if"
  statements had to be converted to use braces.

  * Excessive use of PATH_MAX.  Previous maintainers of this module
  used PATH_MAX to allocate character arrays for filesystem paths
  and SQL statements as though they cost nothing.  In fact, PATH_MAX
  is defined as 4096 bytes!  Some functions had (and still have)
  multiples of these.  One function has 7.  Given that the vast
  majority of installations use the default spool directory path
  `/var/spool/asterisk/voicemail`, the actual path length is usually
  less than 80 bytes.  That's over 4000 bytes wasted.  It was the
  same for SQL statement buffers.  A 4K buffer for statement that
  only needed 60 bytes.  All of these PATH_MAX allocations in the
  ODBC related code were changed to dynamically allocated buffers.
  The rest will have to be addressed separately.

  * Bug fixes.  During the development of this feature, several
  pre-existing ODBC related bugs were discovered and fixed.  They
  had to do with leaving orphaned files on disk, not preserving
  original message ids when moving messages between folders,
  not honoring the "formats" config parameter in certain circumstances,
  etc.

  UserNote: This commit adds a new voicemail.conf option
  'odbc_audio_on_disk' which when set causes the ODBC variant of
  app_voicemail_odbc to leave the message and greeting audio files
  on disk and only store the message metadata in the database.
  Much more information can be found in the voicemail.conf.sample
  file.


#### bridge_basic.c: Make sure that ast_bridge_channel is not destroyed while itera..
  Author: Tinet-mucw
  Date:   2024-06-13

  Resolves: https://github.com/asterisk/asterisk/issues/768

#### app_queue:  Add option to not log Restricted Caller ID to queue_log
  Author: Alexei Gradinari
  Date:   2024-06-12

  Add a queue option log-restricted-caller-id to strip the Caller ID when storing the ENTERQUEUE event
  in the queue log if the Caller ID is restricted.

  Resolves: #765

  UpgradeNote: Add a new column to the queues table:
  queue_log_option_log_restricted ENUM('0','1','off','on','false','true','no','yes')
  to control whether the Restricted Caller ID will be stored in the queue log.

  UserNote: Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID
  will be stored in the queue log.
  If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted.


#### pbx.c: expand fields width of "core show hints"
  Author: Alexei Gradinari
  Date:   2024-06-13

  The current width for "extension" is 20 and "device state id" is 20, which is too small.
  The "extension" field contains "ext"@@"context", so 20 characters is not enough.
  The "device state id" field, for example for Queue pause state contains Queue:"queue_name"_pause_PSJIP/"endpoint", so the 20 characters is not enough.

  Increase the width of "extension" field to 30 characters and the width of the "device state id" field to 60 characters.

  Resolves: #770

  UserNote: The fields width of "core show hints" were increased.
  The width of "extension" field to 30 characters and
  the width of the "device state id" field to 60 characters.


#### pjsip: Add PJSIP_PARSE_URI_FROM dialplan function.
  Author: Sean Bright
  Date:   2024-06-02

  Various SIP headers permit a URI to be prefaced with a `display-name`
  production that can include characters (like commas and parentheses)
  that are problematic for Asterisk's dialplan parser and, specifically
  in the case of this patch, the PJSIP_PARSE_URI function.

  This patch introduces a new function - `PJSIP_PARSE_URI_FROM` - that
  behaves identically to `PJSIP_PARSE_URI` except that the first
  argument is now a variable name and not a literal URI.

  Fixes #756


#### manager.c: Properly terminate `CoreShowChannelMap` event.
  Author: Sean Bright
  Date:   2024-06-10

  Fixes #761


#### xml.c: Update deprecated libxml2 API usage.
  Author: Sean Bright
  Date:   2024-05-23

  Two functions are deprecated as of libxml2 2.12:

    * xmlSubstituteEntitiesDefault
    * xmlParseMemory

  So we update those with supported API.

  Additionally, `res_calendar_caldav` has been updated to use libxml2's
  xmlreader API instead of the SAX2 API which has always felt a little
  hacky (see deleted comment block in `res_calendar_caldav.c`).

  The xmlreader API has been around since libxml2 2.5.0 which was
  released in 2003.

  Fixes #725


#### cdr_pgsql: Fix crash when the module fails to load multiple times.
  Author: chrsmj
  Date:   2024-05-16

  Missing or corrupt cdr_pgsql.conf configuration file can cause the
  second attempt to load the PostgreSQL CDR module to crash Asterisk via
  the Command Line Interface because a null CLI command is registered on
  the first failed attempt to load the module.

  Resolves: #736

#### asterisk.c: Don't log an error if .asterisk_history does not exist.
  Author: Sean Bright
  Date:   2024-05-27

  Fixes #751


#### chan_ooh323: Fix R/0 typo in docs
  Author: Walter Doekes
  Date:   2024-05-27


#### bundled_pjproject: Disable UPnP support.
  Author: Sean Bright
  Date:   2024-05-24

  Fixes #747


#### file.h: Rename function argument to avoid C++ keyword clash.
  Author: Sean Bright
  Date:   2024-05-24

  Fixes #744


#### rtp_engine: add support for multirate RFC2833 digits
  Author: Mike Bradeen
  Date:   2024-04-08

  Add RFC2833 DTMF support for 16K, 24K, and 32K bitrate codecs.

  Asterisk currently treats RFC2833 Digits as a single rtp payload type
  with a fixed bitrate of 8K.  This change would expand that to 8, 16,
  24 and 32K.

  This requires checking the offered rtp types for any of these bitrates
  and then adding an offer for each (if configured for RFC2833.)  DTMF
  generation must also be changed in order to look at the current outbound
  codec in order to generate appropriately timed rtp.

  For cases where no outgoing audio has yet been sent prior to digit
  generation, Asterisk now has a concept of a 'preferred' codec based on
  offer order.

  On inbound calls Asterisk will mimic the payload types of the RFC2833
  digits.

  On outbound calls Asterisk will choose the next free payload types starting
  with 101.

  UserNote: No change in configuration is required in order to enable this
  feature. Endpoints configured to use RFC2833 will automatically have this
  enabled. If the endpoint does not support this, it should not include it in
  the SDP offer/response.

  Resolves: #699
@
text
@d1 1
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# $NetBSD: Makefile,v 1.161 2024/07/08 04:04:52 jnemeth Exp $
d10 1
@


1.161
log
@Update to Asterisk 18.23.1.

## Change Log for Release asterisk-18.23.1

### Summary:

- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9): res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests

### Closed Issues:

  - !GHSA-qqxj-v78h-hrf9: res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests

### Commits By Author:

- ### George Joseph (1):
  - Revert "res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport ad..


-----


## Change Log for Release asterisk-18.23.0

### Summary:

- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0

### User Notes:

- #### res_pjsip_logger: Preserve logging state on reloads.
  Issuing "pjsip reload" will no longer disable
  logging if it was previously enabled from the CLI.

- #### loader.c: Allow dependent modules to be unloaded recursively.
  In certain circumstances, modules with dependency relations
  can have their dependents automatically recursively unloaded and loaded
  again using the "module refresh" CLI command or the ModuleLoad AMI command.

- #### tcptls/iostream:  Add support for setting SNI on client TLS connections
  Secure websocket client connections now send SNI in
  the TLS client hello.

- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
  set identify_by=transport for the pjsip endpoint. Then
  use the existing 'match' option and the new 'transport' option of
  the identify.
  Fixes: #672

- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
  this new feature let users match endpoints based on the
  indound SIP requests' URI. To do so, add 'request_uri' to the
  endpoint's 'identify_by' option. The 'match_request_uri' option of
  the identify can be an exact match for the entire request uri, or a
  regular expression (between slashes). It's quite similar to the
  header identifer.
  Fixes: #599

- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
  the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.

- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
  When using the Originate AMI Action, we now can pass the PreDialGoSub
  parameter, instructing the asterisk to perform an subrouting at
  channel before call start. With this parameter an call initiated
  by AMI can request the channel to start the call automaticaly,
  adding a SIP header to using GoSUB, instructing to autoanswer
  the channel, and proceeding the outbuound extension executing.
  Exemple of an context to perform the previus indication:
  [addautoanswer]
  exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
  exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
  exten => _s,n,Return()

- #### manager.c: Add CLI command to kick AMI sessions.
  The "manager kick session" CLI command now
  allows kicking a specified AMI session.

- #### chan_dahdi: Allow specifying waitfordialtone per call.
  "waitfordialtone" may now be specified for DAHDI
  trunk channels on a per-call basis using the CHANNEL function.

- #### Upgrade bundled pjproject to 2.14.1
  Bundled pjproject has been upgraded to 2.14.1. For more
  information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.14.1


### Upgrade Notes:

- #### pbx_variables.c: Prevent SEGV due to stack overflow.
  The maximum amount of dialplan recursion
  using variable substitution (such as by using EVAL_EXTEN)
  is capped at 15.


## Issue and Commit Detail:

### Closed Issues:

  - 246: [bug]: res_pjsip_logger: Reload disables logging
  - 472: [new-feature]: chan_dahdi: Allow waitfordialtone to be specified per call
  - 474: [new-feature]: loader.c: Allow dependent modules to be unloaded automatically
  - 480: [improvement]: pbx_variables.c: Prevent infinite recursion and stack overflow with variable expansion
  - 485: [new-feature]: manager.c: Allow kicking specific manager sessions
  - 525: [bug]: say.c: Money announcements off by one cent due to floating point rounding
  - 579: [improvement]: Allow GET_TRANSFERRER_DATA for pjsip
  - 599: [improvement]: Endpoint identifier request line
  - 611: [bug]: res_pjsip_session: Polling on non-existing file descriptors when stream is removed
  - 624: [bug]: Park() application does not continue execution if lot is full
  - 642: [bug]: Prometheus bridge metrics contains duplicate entries and help
  - 666: [improvement]: ARI debug should contain endpoint and method
  - 669: [bug]: chan_dahdi: Tens or hundreds of thousands of channel opens attempted during restart
  - 672: [improvement]: Endpoint identifier transport
  - 673: [new-feature]: chan_dahdi: Add AMI action to show spans
  - 676: [bug]: res_stir_shaken implicit declaration of function errors/warnings
  - 681: [new-feature]: callerid.c: Parse all received parameters
  - 683: [improvement]: func_callerid: Warn if invalid redirecting reason is set
  - 689: [bug] Document the `Events` argument of the `Login` AMI action
  - 696: [bug]: Unexpected control subclass '14'
  - 713: [bug]: SNI isn't being set on websocket client connections
  - 716: [bug]: Memory leak in res_stir_shaken tn_config, plus a few other issues
  - 719: [bug]: segfault on start if compiled with DETECT_DEADLOCKS
  - 721: [improvement]: logger: Add unique verbose prefixes for higher verbose levels
  - 729: [bug]: Build failure with uclibc-ng
  - ASTERISK-29912: res_pjsip: module reload disables logging

### Commits By Author:

- ### Fabrice Fontaine (1):
  - res/stasis/control.c: include signal.h

- ### George Joseph (8):
  - Fix incorrect application and function documentation references
  - res_stir_shaken:  Fix compilation for CentOS7 (openssl 1.0.2)
  - manager.c: Add missing parameters to Login documentation
  - rtp_engine and stun: call ast_register_atexit instead of ast_register_cleanup
  - logger.h:  Add SCOPE_CALL and SCOPE_CALL_WITH_RESULT
  - make_buildopts_h: Always include DETECT_DEADLOCKS
  - stir_shaken:  Fix memory leak, typo in config, tn canonicalization
  - tcptls/iostream:  Add support for setting SNI on client TLS connections

- ### Henrik Liljedahl (1):
  - res_pjsip_sdp_rtp.c: Initial RTP inactivity check must consider the rtp_timeou..

- ### Holger Hans Peter Freyther (1):
  - res_prometheus: Fix duplicate output of metric and help text

- ### Ivan Poddubny (2):
  - asterisk.c: Fix sending incorrect messages to systemd notify
  - configs: Fix a misleading IPv6 ACL example in Named ACLs

- ### Joshua Elson (1):
  - Implement Configurable TCP Keepalive Settings in PJSIP Transports

- ### Martin Nystroem (1):
  - res_ari.c: Add additional output to ARI requests when debug is enabled

- ### Martin Tomec (1):
  - res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA

- ### Maximilian Fridrich (1):
  - res_pjsip_session: Reset pending_media_state->read_callbacks

- ### Naveen Albert (14):
  - res_parking: Fail gracefully if parking lot is full.
  - chan_dahdi: Allow specifying waitfordialtone per call.
  - manager.c: Add CLI command to kick AMI sessions.
  - pbx_variables.c: Prevent SEGV due to stack overflow.
  - menuselect: Minor cosmetic fixes.
  - chan_dahdi: Don't retry opening nonexistent channels on restart.
  - chan_dahdi: Add DAHDIShowStatus AMI action.
  - func_callerid: Emit warning if invalid redirecting reason set.
  - file.c, channel.c: Don't emit warnings if progress received.
  - callerid.c: Parse previously ignored Caller ID parameters.
  - loader.c: Allow dependent modules to be unloaded recursively.
  - say.c: Fix cents off-by-one due to floating point rounding.
  - logger: Add unique verbose prefixes for levels 5-10.
  - res_pjsip_logger: Preserve logging state on reloads.

- ### Sean Bright (8):
  - res_monitor.c: Don't emit a warning about 'X' being unrecognized.
  - alembic: Quote new MySQL keyword 'qualify.'
  - res_pjsip: Fix alembic downgrade for boolean columns.
  - res_config_mysql.c: Support hostnames up to 255 bytes.
  - alembic: Fix compatibility with SQLAlchemy 2.0+.
  - cli.c: `core show channels concise` is not really deprecated.
  - alembic: Correct NULLability of PJSIP id columns.
  - app_queue.c: Properly handle invalid strategies from realtime.

- ### Sperl Viktor (2):
  - res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
  - res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.

- ### Spiridonov Dmitry (1):
  - sorcery.c: Fixed crash error when executing "module reload"

- ### Stanislav Abramenkov (1):
  - Upgrade bundled pjproject to 2.14.1

- ### jonatascalebe (1):
  - manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action


### Commit List:

-  configs: Fix a misleading IPv6 ACL example in Named ACLs
-  asterisk.c: Fix sending incorrect messages to systemd notify
-  res/stasis/control.c: include signal.h
-  res_pjsip_logger: Preserve logging state on reloads.
-  logger: Add unique verbose prefixes for levels 5-10.
-  say.c: Fix cents off-by-one due to floating point rounding.
-  loader.c: Allow dependent modules to be unloaded recursively.
-  tcptls/iostream:  Add support for setting SNI on client TLS connections
-  stir_shaken:  Fix memory leak, typo in config, tn canonicalization
-  make_buildopts_h: Always include DETECT_DEADLOCKS
-  sorcery.c: Fixed crash error when executing "module reload"
-  callerid.c: Parse previously ignored Caller ID parameters.
-  logger.h:  Add SCOPE_CALL and SCOPE_CALL_WITH_RESULT
-  app_queue.c: Properly handle invalid strategies from realtime.
-  file.c, channel.c: Don't emit warnings if progress received.
-  alembic: Correct NULLability of PJSIP id columns.
-  rtp_engine and stun: call ast_register_atexit instead of ast_register_cleanup
-  manager.c: Add missing parameters to Login documentation
-  func_callerid: Emit warning if invalid redirecting reason set.
-  chan_dahdi: Add DAHDIShowStatus AMI action.
-  res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-  res_stir_shaken:  Fix compilation for CentOS7 (openssl 1.0.2)
-  Fix incorrect application and function documentation references
-  cli.c: `core show channels concise` is not really deprecated.
-  res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-  Implement Configurable TCP Keepalive Settings in PJSIP Transports
-  chan_dahdi: Don't retry opening nonexistent channels on restart.
-  res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-  res_ari.c: Add additional output to ARI requests when debug is enabled
-  alembic: Fix compatibility with SQLAlchemy 2.0+.
-  manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-  menuselect: Minor cosmetic fixes.
-  pbx_variables.c: Prevent SEGV due to stack overflow.
-  res_prometheus: Fix duplicate output of metric and help text
-  manager.c: Add CLI command to kick AMI sessions.
-  chan_dahdi: Allow specifying waitfordialtone per call.
-  res_parking: Fail gracefully if parking lot is full.
-  res_config_mysql.c: Support hostnames up to 255 bytes.
-  res_pjsip: Fix alembic downgrade for boolean columns.
-  Upgrade bundled pjproject to 2.14.1
-  alembic: Quote new MySQL keyword 'qualify.'
-  res_pjsip_session: Reset pending_media_state->read_callbacks
-  res_monitor.c: Don't emit a warning about 'X' being unrecognized.

### Commit Details:

#### configs: Fix a misleading IPv6 ACL example in Named ACLs
  Author: Ivan Poddubny
  Date:   2024-05-05

  "deny=::" is equivalent to "::/128".
  In order to mean "deny everything by default" it must be "::/0".


#### asterisk.c: Fix sending incorrect messages to systemd notify
  Author: Ivan Poddubny
  Date:   2024-05-05

  Send "RELOADING=1" instead of "RELOAD=1" to follow the format
  expected by systemd (see sd_notify(3) man page).

  Do not send STOPPING=1 in remote console mode:
  attempting to execute "asterisk -rx" by the main process leads to
  a warning if NotifyAccess=main (the default) or to a forced termination
  if NotifyAccess=all.


#### res/stasis/control.c: include signal.h
  Author: Fabrice Fontaine
  Date:   2024-05-01

  Include signal.h to avoid the following build failure with uclibc-ng
  raised since
  https://github.com/asterisk/asterisk/commit/2694792e13c7f3ab1911c4a69fba0df32c544177:

  stasis/control.c: In function 'exec_command_on_condition':
  stasis/control.c:313:3: warning: implicit declaration of function 'pthread_kill'; did you mean 'pthread_yield'? [-Wimplicit-function-declaration]
    313 |   pthread_kill(control->control_thread, SIGURG);
        |   ^~~~~~~~~~~~
        |   pthread_yield
  stasis/control.c:313:41: error: 'SIGURG' undeclared (first use in this function)
    313 |   pthread_kill(control->control_thread, SIGURG);
        |                                         ^~~~~~

  cherry-pick-to: 18
  cherry-pick-to: 20
  cherry-pick-to: 21

  Fixes: #729

#### res_pjsip_logger: Preserve logging state on reloads.
  Author: Naveen Albert
  Date:   2023-08-09

  Currently, reloading res_pjsip will cause logging
  to be disabled. This is because logging can also
  be controlled via the debug option in pjsip.conf
  and this defaults to "no".

  To improve this, logging is no longer disabled on
  reloads if logging had not been previously
  enabled using the debug option from the config.
  This ensures that logging enabled from the CLI
  will persist through a reload.

  ASTERISK-29912 #close

  Resolves: #246

  UserNote: Issuing "pjsip reload" will no longer disable
  logging if it was previously enabled from the CLI.


#### logger: Add unique verbose prefixes for levels 5-10.
  Author: Naveen Albert
  Date:   2024-04-27

  Add unique verbose prefixes for levels higher than 4, so
  that these can be visually differentiated from each other.

  Resolves: #721

#### say.c: Fix cents off-by-one due to floating point rounding.
  Author: Naveen Albert
  Date:   2024-01-10

  Some of the money announcements can be off by one cent,
  due to the use of floating point in the money calculations,
  which is bad for obvious reasons.

  This replaces floating point with simple string parsing
  to ensure the cents value is converted accurately.

  Resolves: #525

#### loader.c: Allow dependent modules to be unloaded recursively.
  Author: Naveen Albert
  Date:   2023-12-02

  Because of the (often recursive) nature of module dependencies in
  Asterisk, hot swapping a module on the fly is cumbersome if a module
  is depended on by other modules. Currently, dependencies must be
  popped manually by unloading dependents, unloading the module of
  interest, and then loading modules again in reverse order.

  To make this easier, the ability to do this recursively in certain
  circumstances has been added, as an optional extension to the
  "module refresh" command. If requested, Asterisk will check if a module
  that has a positive usecount could be unloaded safely if anything
  recursively dependent on it were unloaded. If so, it will go ahead
  and unload all these modules and load them back again. This makes
  hot swapping modules that provide dependencies much easier.

  Resolves: #474

  UserNote: In certain circumstances, modules with dependency relations
  can have their dependents automatically recursively unloaded and loaded
  again using the "module refresh" CLI command or the ModuleLoad AMI command.


#### res_pjsip_sdp_rtp.c: Initial RTP inactivity check must consider the rtp_timeou..
  Author: Henrik Liljedahl
  Date:   2024-04-11

  First rtp activity check was performed after 500ms regardless of the rtp_timeout setting. Having a call in ringing state for more than rtp_timeout and the first rtp package is received more than 500ms after sdp negotiation and before the rtp_timeout, erronously caused the call to be hungup. Changed to perform the first rtp inactivity check after the timeout setting preventing calls to be disconnected before the rtp_timeout has elapsed since sdp negotiation.

  Fixes #710


#### tcptls/iostream:  Add support for setting SNI on client TLS connections
  Author: George Joseph
  Date:   2024-04-23

  If the hostname field of the ast_tcptls_session_args structure is
  set (which it is for websocket client connections), that hostname
  will now automatically be used in an SNI TLS extension in the client
  hello.

  Resolves: #713

  UserNote: Secure websocket client connections now send SNI in
  the TLS client hello.


#### stir_shaken:  Fix memory leak, typo in config, tn canonicalization
  Author: George Joseph
  Date:   2024-04-25

  * Fixed possible memory leak in tn_config:tn_get_etn() where we
  weren't releasing etn if tn or eprofile were null.
  * We now canonicalize TNs before using them for lookups or adding
  them to Identity headers.
  * Fixed a typo in stir_shaken.conf.sample.

  Resolves: #716

#### make_buildopts_h: Always include DETECT_DEADLOCKS
  Author: George Joseph
  Date:   2024-04-27

  Since DETECT_DEADLOCKS is now split from DEBUG_THREADS, it must
  always be included in buildopts.h instead of only when
  ADD_CFLAGS_TO_BUILDOPTS_H is defined.  A SEGV will result otherwise.

  Resolves: #719

#### sorcery.c: Fixed crash error when executing "module reload"
  Author: Spiridonov Dmitry
  Date:   2024-04-14

  Fixed crash error when cli "module reload". The error appears when
  compiling with res_prometheus and using the sorcery memory cache for
  registrations


#### callerid.c: Parse previously ignored Caller ID parameters.
  Author: Naveen Albert
  Date:   2024-04-01

  Commit f2f397c1a8cc48913434ebb297f0ff50d96993db previously
  made it possible to send Caller ID parameters to FXS stations
  which, prior to that, could not be sent.

  This change is complementary in that we now handle receiving
  all these parameters on FXO lines and provide these up to
  the dialplan, via chan_dahdi. In particular:

  * If a redirecting reason is provided, the channel's redirecting
    reason is set. No redirecting number is set, since there is
    no parameter for this in the Caller ID protocol, but the reason
    can be checked to determine if and why a call was forwarded.
  * If the Call Qualifier parameter is received, the Call Qualifier
    variable is set.
  * Some comments have been added to explain why some of the code
    is the way it is, to assist other people looking at it.

  With this change, Asterisk's Caller ID implementation is now
  reasonably complete for both FXS and FXO operation.

  Resolves: #681

#### logger.h:  Add SCOPE_CALL and SCOPE_CALL_WITH_RESULT
  Author: George Joseph
  Date:   2024-04-09

  If you're tracing a large function that may call another function
  multiple times in different circumstances, it can be difficult to
  see from the trace output exactly which location that function
  was called from.  There's no good way to automatically determine
  the calling location.  SCOPE_CALL and SCOPE_CALL_WITH_RESULT
  simply print out a trace line before and after the call.

  The difference between SCOPE_CALL and SCOPE_CALL_WITH_RESULT is
  that SCOPE_CALL ignores the function's return value (if any) where
  SCOPE_CALL_WITH_RESULT allows you to specify the type of the
  function's return value so it can be assigned to a variable.
  SCOPE_CALL_WITH_INT_RESULT is just a wrapper for SCOPE_CALL_WITH_RESULT
  and the "int" return type.


#### app_queue.c: Properly handle invalid strategies from realtime.
  Author: Sean Bright
  Date:   2024-04-13

  The existing code sets the queue strategy to `ringall` but it is then
  immediately overwritten with an invalid one.

  Fixes #707


#### file.c, channel.c: Don't emit warnings if progress received.
  Author: Naveen Albert
  Date:   2024-04-09

  Silently ignore AST_CONTROL_PROGRESS where appropriate,
  as most control frames already are.

  Resolves: #696

#### alembic: Correct NULLability of PJSIP id columns.
  Author: Sean Bright
  Date:   2024-04-06

  Fixes #695


#### rtp_engine and stun: call ast_register_atexit instead of ast_register_cleanup
  Author: George Joseph
  Date:   2024-04-02

  rtp_engine.c and stun.c were calling ast_register_cleanup which
  is skipped if any loadable module can't be cleanly unloaded
  when asterisk shuts down.  Since this will always be the case,
  their cleanup functions never get run.  In a practical sense
  this makes no difference since asterisk is shutting down but if
  you're in development mode and trying to use the leak sanitizer,
  the leaks from both of those modules clutter up the output.


#### manager.c: Add missing parameters to Login documentation
  Author: George Joseph
  Date:   2024-04-03

  * Added the AuthType and Key parameters for MD5 authentication.

  * Added the Events parameter.

  Resolves: #689

#### func_callerid: Emit warning if invalid redirecting reason set.
  Author: Naveen Albert
  Date:   2024-04-01

  Emit a warning if REDIRECTING(reason) is set to an invalid
  reason, consistent with what happens when
  REDIRECTING(orig-reason) is set to an invalid reason.

  Resolves: #683

#### chan_dahdi: Add DAHDIShowStatus AMI action.
  Author: Naveen Albert
  Date:   2024-03-29

  * Add an AMI action to correspond to the "dahdi show status"
    command, allowing span information to be retrieved via AMI.
  * Show span number and sig type in "dahdi show channels".

  Resolves: #673

#### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
  Author: Sperl Viktor
  Date:   2024-03-28

  Add a new identify_by option to res_pjsip_endpoint_identifier_ip
  called 'transport' this matches endpoints based on the bound
  ip address (local) instead of the 'ip' option, which matches on
  the source ip address (remote).

  UserNote: set identify_by=transport for the pjsip endpoint. Then
  use the existing 'match' option and the new 'transport' option of
  the identify.

  Fixes: #672

#### res_stir_shaken:  Fix compilation for CentOS7 (openssl 1.0.2)
  Author: George Joseph
  Date:   2024-04-01

  * OpenSSL 1.0.2 doesn't support X509_get0_pubkey so we now use
    X509_get_pubkey.  The difference is that X509_get_pubkey requires
    the caller to free the EVP_PKEY themselves so we now let
    RAII_VAR do that.
  * OpenSSL 1.0.2 doesn't support upreffing an X509_STORE so we now
    wrap it in an ao2 object.
  * OpenSSL 1.0.2 doesn't support X509_STORE_get0_objects to get all
    the certs from an X509_STORE and there's no easy way to polyfill
    it so the CLI commands that list profiles will show a "not
    supported" message instead of listing the certs in a store.

  Resolves: #676

#### Fix incorrect application and function documentation references
  Author: George Joseph
  Date:   2024-04-01

  There were a few references in the embedded documentation XML
  where the case didn't match or where the referenced app or function
  simply didn't exist any more.  These were causing 404 responses
  in docs.asterisk.org.


#### cli.c: `core show channels concise` is not really deprecated.
  Author: Sean Bright
  Date:   2024-04-01

  Fixes #675


#### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
  Author: Sperl Viktor
  Date:   2024-03-28

  Add ability to match against PJSIP request URI.

  UserNote: this new feature let users match endpoints based on the
  indound SIP requests' URI. To do so, add 'request_uri' to the
  endpoint's 'identify_by' option. The 'match_request_uri' option of
  the identify can be an exact match for the entire request uri, or a
  regular expression (between slashes). It's quite similar to the
  header identifer.

  Fixes: #599

#### Implement Configurable TCP Keepalive Settings in PJSIP Transports
  Author: Joshua Elson
  Date:   2024-03-18

  This commit introduces configurable TCP keepalive settings for
  both TCP and TLS transports. The changes allow for finer control
  over TCP connection keepalives, enhancing stability and reliability
  in environments prone to connection timeouts or where intermediate
  devices may prematurely close idle connections. This has proven
  necessary and has already been tested in production in several
  specialized environments where access to the underlying transport
  is unreliable in ways invisible to the operating system directly,
  so these keepalive and timeout mechanisms are necessary.

  Fixes #657


#### chan_dahdi: Don't retry opening nonexistent channels on restart.
  Author: Naveen Albert
  Date:   2024-03-26

  Commit 729cb1d390b136ccc696430aa5c68d60ea4028be added logic to retry
  opening DAHDI channels on "dahdi restart" if they failed initially,
  up to 1,000 times in a loop, to address cases where the channel was
  still in use. However, this retry loop does not use the actual error,
  which means chan_dahdi will also retry opening nonexistent channels
  1,000 times per channel, causing a flood of unnecessary warning logs
  for an operation that will never succeed, with tens or hundreds of
  thousands of open attempts being made.

  The original patch would have been more targeted if it only retried
  on the specific relevant error (likely EBUSY, although it's hard to
  say since the original issue is no longer available).

  To avoid the problem above while avoiding the possibility of breakage,
  this skips the retry logic if the error is ENXIO (No such device or
  address), since this will never succeed.

  Resolves: #669

#### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
  Author: Martin Tomec
  Date:   2024-02-06

  There was functionality in chan_sip to get REFER headers, with
  GET_TRANSFERRER_DATA variable. This commit implements the same
  functionality in pjsip, to ease transfer from chan_sip to pjsip.

  Fixes: #579

  UserNote: the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.

#### res_ari.c: Add additional output to ARI requests when debug is enabled
  Author: Martin Nystroem
  Date:   2024-03-22

  When ARI debug is enabled the logs will now output http method and the uri.

  Fixes: #666

#### alembic: Fix compatibility with SQLAlchemy 2.0+.
  Author: Sean Bright
  Date:   2024-03-20

  SQLAlchemy 2.0 changed the way that commits/rollbacks are handled
  causing the final `UPDATE` to our `alembic_version_<whatever>` tables
  to be rolled back instead of committed.

  We now use one connection to determine which
  `alembic_version_<whatever>` table to use and another to run the
  actual migrations. This prevents the erroneous rollback.

  This change is compatible with both SQLAlchemy 1.4 and 2.0.


#### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
  Author: jonatascalebe
  Date:   2024-03-14

  manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action

  The action originate does not has the ability to run an subroutine
  at initial channel, like the Aplication Originate. This update
  give this ability for de action originate too.

  For example, we can run a routine via Gosub on the channel to
  request an automatic answer, so the caller does not need to accept
  the call when using the originate command via manager, making
  the operation more efficient.

  UserNote: When using the Originate AMI Action, we now can pass
  the PreDialGoSub parameter, instructing the asterisk to perform
  an subrouting at channel before call start. With this parameter
  an call initiated by AMI can request the channel to start the
  call automaticaly, adding a SIP header to using GoSUB, instructing
  to autoanswer the channel, and proceeding the outbuound extension
  executing. Exemple of an context to perform the previus indication:
  [addautoanswer]
  exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
  exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
  exten => _s,n,Return()


#### menuselect: Minor cosmetic fixes.
  Author: Naveen Albert
  Date:   2024-03-21

  Improve some of the formatting from
  dd3f17c699e320d6d30c94298d8db49573ba28da
  (#521).


#### pbx_variables.c: Prevent SEGV due to stack overflow.
  Author: Naveen Albert
  Date:   2024-03-21

  It is possible for dialplan to result in an infinite
  recursion of variable substitution, which eventually
  leads to stack overflow. If we detect this, abort
  substitution and log an error for the user to fix
  the broken dialplan.

  Resolves: #480

  UpgradeNote: The maximum amount of dialplan recursion
  using variable substitution (such as by using EVAL_EXTEN)
  is capped at 15.


#### res_prometheus: Fix duplicate output of metric and help text
  Author: Holger Hans Peter Freyther
  Date:   2024-02-24

  The prometheus exposition format requires each line to be unique[1].
  This is handled by struct prometheus_metric having a list of children
  that is managed when registering a metric. In case the scrape callback
  is used, it is the responsibility of the implementation to handle this
  correctly.

  Originally the bridge callback didn't handle NULL snapshots, the crash
  fix lead to NULL metrics, and fixing that lead to duplicates.

  The original code assumed that snapshots are not NULL and then relied on
  "if (i > 0)" to establish the parent/children relationship between
  metrics of the same class. This is not workerable as the first bridge
  might be invisible/lacks a snapshot.

  Fix this by keeping a separate array of the first metric by class.
  Instead of relying on the index of the bridge, check whether the array
  has an entry. Use that array for the output.

  Add a test case that verifies that the help text is not duplicated.

  Resolves: #642

  [1] https://prometheus.io/docs/instrumenting/exposition_formats/#grouping-and-sorting


#### manager.c: Add CLI command to kick AMI sessions.
  Author: Naveen Albert
  Date:   2023-12-06

  This adds a CLI command that can be used to manually
  kick specific AMI sessions.

  Resolves: #485

  UserNote: The "manager kick session" CLI command now
  allows kicking a specified AMI session.


#### chan_dahdi: Allow specifying waitfordialtone per call.
  Author: Naveen Albert
  Date:   2023-12-02

  The existing "waitfordialtone" setting in chan_dahdi.conf
  applies permanently to a specific channel, regardless of
  how it is being used. This rather restrictively prevents
  a system from simultaneously being able to pick free lines
  for outgoing calls while also allowing barge-in to a trunk
  by some other arrangement.

  This allows specifying "waitfordialtone" using the CHANNEL
  function for only the next call that will be placed, allowing
  significantly more flexibility in the use of trunk interfaces.

  Resolves: #472

  UserNote: "waitfordialtone" may now be specified for DAHDI
  trunk channels on a per-call basis using the CHANNEL function.


#### res_parking: Fail gracefully if parking lot is full.
  Author: Naveen Albert
  Date:   2024-03-03

  Currently, if a parking lot is full, bridge setup returns -1,
  causing dialplan execution to terminate without TryExec.
  However, such failures should be handled more gracefully,
  the same way they are on other paths, as indicated by the
  module's author, here:

  http://lists.digium.com/pipermail/asterisk-dev/2018-December/077144.html

  Now, callers will hear the parking failure announcement, and dialplan
  will continue, which is consistent with existing failure modes.

  Resolves: #624

#### res_config_mysql.c: Support hostnames up to 255 bytes.
  Author: Sean Bright
  Date:   2024-03-18

  Fixes #654


#### res_pjsip: Fix alembic downgrade for boolean columns.
  Author: Sean Bright
  Date:   2024-03-18

  When downgrading, ensure that we don't touch columns that didn't
  actually change during upgrade.


#### Upgrade bundled pjproject to 2.14.1
  Author: Stanislav Abramenkov
  Date:   2024-03-12

  Fixes: asterisk#648

  UserNote: Bundled pjproject has been upgraded to 2.14.1. For more
  information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.14.1


#### alembic: Quote new MySQL keyword 'qualify.'
  Author: Sean Bright
  Date:   2024-03-15

  Fixes #651


#### res_pjsip_session: Reset pending_media_state->read_callbacks
  Author: Maximilian Fridrich
  Date:   2024-02-15

  In handle_negotiated_sdp the pending_media_state->read_callbacks must be
  reset before they are added in the SDP handlers in
  handle_negotiated_sdp_session_media. Otherwise, old callbacks for
  removed streams and file descriptors could be added to the channel and
  Asterisk would poll on non-existing file descriptors.

  Resolves: #611

#### res_monitor.c: Don't emit a warning about 'X' being unrecognized.
  Author: Sean Bright
  Date:   2024-03-07

  Code was added in 030f7d41 to warn if an unrecognized option was
  passed to an application, but code in Monitor was taking advantage of
  the fact that the application would silently accept an invalid option.

  We now recognize the invalid option but we don't do anything if it's
  set.

  Fixes #639
@
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DISTNAME=	asterisk-18.23.1
d311 5
@


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@revbump after icu and protobuf updates
@
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# $NetBSD: Makefile,v 1.159 2024/05/20 14:51:16 jnemeth Exp $
d9 1
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DISTNAME=	asterisk-18.22.0
#PKGREVISION=	24
PKGREVISION=	2
d164 1
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PJPROJ_VERSION=	2.14
d308 3
@


1.159
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@switch to https
@
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PKGREVISION=	1
@


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log
@*: recursive bump for gnutls p11-kit option

(existing installations need the bl3.mk included, but it's now only
optionally included)
@
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MASTER_SITES=	http://downloads.asterisk.org/pub/telephony/asterisk/
MASTER_SITES+=	http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/
MASTER_SITES+=	http://downloads.asterisk.org/pub/telephony/sounds/releases/
@


1.157
log
@Update to Asterisk 18.22.0:

pkgsrc changes
- adapt for newer version of pjsip
- adapt for new dependency (required for STIR/SHAKEN):  libjwt

Change Log for Release asterisk-18.22.0
========================================

Summary:
----------------------------------------

- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(â€¦) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.

User Notes:
----------------------------------------

- ### app_dial: Add dial time for progress/ringing.
  The timeout argument to Dial now allows
  specifying the maximum amount of time to dial if
  early media is not received.

- ### app_voicemail: Allow preventing mark messages as urgent.
  The leaveurgent mailbox option can now be used to
  control whether callers may leave messages marked as 'Urgent'.

- ### Stir/Shaken Refactor
  Asterisk's stir-shaken feature has been refactored to
  correct interoperability, RFC compliance, and performance issues.
  See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
  information.

- ### Upgrade bundled pjproject to 2.14.
  Bundled pjproject has been upgraded to 2.14. For more
  information on what all is included in this change, check out the
  pjproject Github page: https://github.com/pjsip/pjproject/releases

- ### app_speech_utils.c: Allow partial speech results.
  The SpeechBackground dialplan application now supports a 'p'
  option that will return partial results from speech engines that
  provide them when a timeout occurs.

- ### app_chanspy: Add 'D' option for dual-channel audio
  The ChanSpy application now accepts the 'D' option which
  will interleave the spied audio within the outgoing frames. The
  purpose of this is to allow the audio to be read as a Dual channel
  stream with separate incoming and outgoing audio. Setting both the
  'o' option and the 'D' option and results in the 'D' option being
  ignored.

- ### chan_dahdi: Allow MWI to be manually toggled on channels.
  The 'dahdi set mwi' now allows MWI on channels
  to be manually toggled if needed for troubleshooting.
  Resolves: #440


Upgrade Notes:
----------------------------------------

- ### Stir/Shaken Refactor
  The stir-shaken refactor is a breaking change but since
  it's not working now we don't think it matters. The
  stir_shaken.conf file has changed significantly which means that
  existing ones WILL need to be changed.  The stir_shaken.conf.sample
  file in configs/samples/ has quite a bit more information.  This is
  also an ABI breaking change since some of the existing objects
  needed to be changed or removed, and new ones added.  Additionally,
  if res_stir_shaken is enabled in menuselect, you'll need to either
  have the development package for libjwt v1.15.3 installed or use
  the --with-libjwt-bundled option with ./configure.
@
text
@d1 1
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d11 1
@


1.156
log
@Update to Asterisk 18.21.0

pkgsrc changes:
- adapt to various upstream changes
- update for newer version of pjproject
- add unconditional depeendency on SDL
- remove pktccops and mgcp option (has to do with supporting cable headends)
- remove various 64-bit time_t fixes as upstream is finally doing these


                               [asterisk-announce] asterisk release 18.21.0


The Asterisk Development Team would like to announce
the release of asterisk-18.21.0.

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!


Change Log for Release asterisk-18.21.0
========================================

Summary:
----------------------------------------

- logger: Fix linking regression.
- Revert "core & res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version >= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work

User Notes:
----------------------------------------

- ### app_dial: Add option "j" to preserve initial stream topology of caller
  The option "j" is now available for the Dial application which
  uses the initial stream topology of the caller to create the outgoing
  channels.

- ### logger: Add channel-based filtering.
  The console log can now be filtered by
  channels or groups of channels, using the
  logger filter CLI commands.

- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
  A new dialplan app PJSIPHangup and AMI action allows you
  to hang up an unanswered incoming PJSIP call with a specific SIP
  response code in the 400 -> 699 range.

- ### app_voicemail: Add AMI event for mailbox PIN changes.
  The VoicemailPasswordChange event is
  now emitted whenever a mailbox password is updated,
  containing the mailbox information and the new
  password.
  Resolves: #398

- ### res_speech: allow speech to translate input channel
  res_speech now supports translation of an input channel
  to a format supported by the speech provider, provided a translation
  path is available between the source format and provider capabilites.

- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
  With this update, the PJSIP realm lengths have been extended
  to support up to 255 characters.

- ### res_stasis: signal when new command is queued
  Call setup times should be significantly improved
  when using ARI.

- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
  You no longer need to select DEBUG_THREADS to use
  DETECT_DEADLOCKS.  This removes a significant amount of overhead
  if you just want to detect possible deadlocks vs needing full
  lock tracing.

- ### file.c: Add ability to search custom dir for sounds
  A new option "sounds_search_custom_dir" has been added to
  asterisk.conf that allows asterisk to search
  AST_DATA_DIR/sounds/custom for sounds files before searching the
  standard AST_DATA_DIR/sounds/<lang> directory.

- ### make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
  The "Build Options" entry in the "core show settings"
  CLI command has been renamed to "ABI related Build Options" and
  a new entry named "All Build Options" has been added that shows
  both breaking and non-breaking options.

- ### chan_rtp: Implement RTP glue for UnicastRTP channels
  The dial string option 'g' was added to the UnicastRTP channel
  which enables RTP glue and therefore native RTP bridges with those
  channels.

- ### app_queue: periodic announcement configurable start time.
  Introduce a new queue configuration option called
  'periodic-announce-startdelay' which will vary the normal (historic)
  behavior of starting the periodic announcement cycle at
  periodic-announce-frequency seconds after entering the queue to start
  the periodic announcement cycle at period-announce-startdelay seconds
  after joining the queue.  The default behavior if this config option is
  not set remains unchanged.
  Signed-off-by: Jaco Kroon <jaco at uls.co.za>

- ### variables: Add additional variable dialplan functions.
  Four new dialplan functions have been added.
  GLOBAL_DELETE and DELETE have been added which allows
  the deletion of global and channel variables.
  GLOBAL_EXISTS and VARIABLE_EXISTS have been added
  which checks whether a global or channel variable has
  been set.


Upgrade Notes:
----------------------------------------

- ### app.c: Allow ampersands in playback lists to be escaped.
  Ampersands in URLs passed to the `Playback()`,
  `Background()`, `SpeechBackground()`, `Read()`, `Authenticate()`, or
  `Queue()` applications as filename arguments can now be escaped by
  single quoting the filename. Additionally, this is also possible when
  using the `CONFBRIDGE` dialplan function, or configuring various
  features in `confbridge.conf` and `queues.conf`.

- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
  The dtls_rekey will be disabled if webrtc support is
  requested on an endpoint. A warning will also be emitted.

- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
  As part of this update, the maximum allowable length
  for PJSIP endpoints and relevant resources has been increased from
  40 to 255 characters. To take advantage of this enhancement, it is
  recommended to run the necessary procedures (e.g., Alembic) to
  update your schemas.


                               [asterisk-announce] asterisk release 18.20.2

The Asterisk Development Team would like to announce
the release of asterisk-18.20.2.

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!


Change Log for Release asterisk-18.20.2
========================================

Summary:
----------------------------------------

- res_rtp_asterisk: Fix regression issues with DTLS client check


                               [asterisk-announce] asterisk release 18.20.1


The Asterisk Development Team would like to announce security release
Asterisk 18.20.1.

The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files](https://github.com/asterisk/asterisk/s
ecurity/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call initiation](https://github
.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ](https://github.com/asterisk/asteris
k/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using 'update'](https://github.com
/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)


Change Log for Release asterisk-18.20.1
========================================

Summary:
----------------------------------------

- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.


                               [asterisk-announce] asterisk release 18.20.0

The Asterisk Development Team would like to announce
the release of asterisk-18.20.0.

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!


Change Log for Release asterisk-18.20.0
========================================

Summary:
----------------------------------------

- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual
  messages in a particular mailbox folder. The forward command can
  be used to copy a message within a mailbox or to another mailbox.
  Also adds a VoicemailBoxSummarry, required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using
  `rtp->themssrc_valid` into the scope of the rtp_i nstance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.

User Notes:
----------------------------------------

- ### sig_analog: Add Called Subscriber Held capability.
  Called Subscriber Held is now supported for analog
  FXS channels, using the calledsubscriberheld option. This allows
  a station  user to go on hook when receiving an incoming call
  and resume from another phone on the same line by going on hook,
  without disconnecting the call.

- ### res_pjsip_header_funcs: Make prefix argument optional.
  The prefix argument to PJSIP_HEADERS is now
  optional. If not specified, all header names will be
  returned.

- ### core/ari/pjsip: Add refer mechanism
  There is a new ARI endpoint `/endpoints/refer` for referring
  an endpoint to some URI or endpoint.

- ### chan_dahdi: Allow autoreoriginating after hangup.
  The autoreoriginate setting now allows for kewlstart FXS
  channels to automatically reoriginate and provide dial tone to the
  user again after all calls on the line have cleared. This saves users
  from having to manually hang up and pick up the receiver again before
  making another call.

- ### sig_analog: Allow three-way flash to time out to silence.
  The threewaysilenthold option now allows the three-way
  dial tone to time out to silence, rather than continuing forever.

- ### res_pjsip: Enable TLS v1.3 if present.
  res_pjsip now allows TLS v1.3 to be enabled if supported by
  the underlying PJSIP library. The bundled version of PJSIP supports
  TLS v1.3.

- ### app_queue: Add support for applying caller priority change immediately.
  The 'queue priority caller' CLI command and
  'QueueChangePriorityCaller' AMI action now have an 'immediate'
  argument which allows the caller priority change to be reflected
  immediately, causing the position of a caller to move within the
  queue depending on the priorities of the other callers.

- ### Adds manager actions to allow move/remove/forward individual
    messages in a particular mailbox folder.  The forward command
    can be used to copy a message within a mailbox or to another
    mailbox. Also adds a Vo icemailBoxSummarry, required to retrieve
    message ID's.
  The following manager actions have been added
    VoicemailBoxSummary - Generate message list for a given mailbox
    VoicemailRemove - Remove a message from a mailbox folder
    VoicemailMove - Move a message from one folder to another within a mailbox
    VoicemailForward - Copy a message from one folder in one mailbox
      to another folder in another or the same mailbox.

- ### app_voicemail: add CLI commands for message manipulation
  The following CLI commands have been added to app_voicemail
  voicemail show mailbox <mailbox> <context>
  Show contents of mailbox <mailbox>@@<context>
  voicemail remove <mailbox> <context> <from_folder> <messageid>
  Remove message <messageid> from <from_folder> in mailbox <mailbox>@@<context>
  voicemail move <mailbox> <context> <from_folder> <messageid> <to_folder>
  Move message <messageid> in mailbox <mailbox>&<context> from <from_folder> to <to_folder>
  voicemail forward <from_mailbox> <from_context> <from_folder>
    <messageid> <to_mailbox> <to_context> <to_ folder>
  Forward message <messageid> in mailbox <mailbox>@@<context> <from_folder> to
  mailbox <mailbox>@@<context> <to_folder>

- ### sig_analog: Allow immediate fake ring to be suppressed.
  The immediatering option can now be set to no to suppress
  the fake audible ringback provided when immediate=yes on FXS channels.


                               [asterisk-announce] Asterisk Release 18.19.0

The Asterisk Development Team would like to announce
the release of Asterisk 18.19.0.

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!


Change Log for Release 18.19.0
========================================

Summary:
----------------------------------------

- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.

User Notes:
----------------------------------------

- ### AMI: Add parking position parameter to Park action
  New ParkingSpace parameter has been added to AMI action Park.

- ### res_musiconhold: Add option to loop last file.
  The loop_last option in musiconhold.conf now
  allows the last file in the directory to be looped once reached.

- ### AMI: Add CoreShowChannelMap action.
  New AMI action CoreShowChannelMap has been added.

- ### sig_analog: Add fuller Caller ID support.
  Additional Caller ID properties are now supported on
  incoming calls to FXS stations, namely the
  redirecting reason and call qualifier.

- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
  When creating a bridge using the ARI the 'type' argument now
  accepts a new value 'sdp_label' which will configure the bridge to add
  labels for each stream in the SDP with the corresponding channel id.

- ### app_queue: Preserve reason for realtime queues
  Make paused reason in realtime queues persist an
  Asterisk restart. This was fixed for non-realtime
  queues in ASTERISK_25732.


Upgrade Notes:
----------------------------------------

- ### app_queue: Preserve reason for realtime queues
  Add a new column to the queue_member table:
  reason_paused VARCHAR(80) so the reason can be preserved.


Closed Issues:
----------------------------------------

  - #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is insufficient/ineffective
  - #55: [bug]: res_sorcery_memory_cache: Memory leak when calling sorcery_memory_cache_open
  - #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
  - #65: [bug]: heap overflow by default at startup
  - #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for realtime queues
  - #73: [new-feature]: pjsip: Allow topology/session refreshes in early media state
  - #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
  - #89: [improvement]:  indications: logging changes
  - #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP labels
  - #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
  - #98: [new-feature]: callerid: Allow timezone to be specified at runtime
  - #100: [bug]: sig_analog: hidecallerid setting is broken
  - #102: [bug]: Strange warning - 'T' option is not compatible with remote console mode and has no effect
.
  - #104: [improvement]: Add AMI action to get a list of connected channels
  - #108: [new-feature]: fair handling of calls in multi-queue scenarios
  - #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
  - #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
  - #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
  - #122: [new-feature]: res_musiconhold: Add looplast option
  - #133: [bug]: unlock channel after moh state access
  - #136: [bug]: Makefile downloader does not follow redirects.
  - #145: [bug]: ABI issue with pjproject and pjsip_inv_session
  - #155: [bug]: GCC 13 is catching a few new trivial issues
  - #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
  - #174: [bug]: app_voicemail imap compile errors
  - #200: [bug]: Regression: In app.h an enum is used before its declaration.


                               [asterisk-announce] Asterisk Release 18.18.1

The Asterisk Development Team would like to announce security release
Asterisk 18.18.1.

The following security advisories were resolved in this release:
https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm


Change Log for Release 18.18.1
========================================

Summary:
----------------------------------------

- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1

User Notes:
----------------------------------------

- ### res_http_media_cache: Introduce options and customize
  The res_http_media_cache module now attempts to load
  configuration from the res_http_media_cache.conf file.
  The following options were added:
    * timeout_secs
    * user_agent
    * follow_location
    * max_redirects
    * protocols
    * redirect_protocols
    * dns_cache_timeout_secs

- ### format_sln: add .slin as supported file extension
  format_sln now recognizes '.slin' as a valid
  file extension in addition to the existing
  '.sln' and '.raw'.

- ### bridge_builtin_features: add beep via touch variable
  Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
  Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
  interval in seconds will result in a periodic beep being
  played to the monitored channel upon MixMontior/Monitor
  feature start.
  If an interval less than 5 seconds is specified, the interval
  will default to 5 seconds.  If the value is set to an invalid
  interval, the default of 15 seconds will be used.

- ### app_senddtmf: Add SendFlash AMI action.
  The SendFlash AMI action now allows sending
  a hook flash event on a channel.

- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
  It is now possible to specify the MixMonitorID when calling
  the manager action: MixMonitorMute.  This will allow an
  individual MixMonitor instance to be muted via ID.
  The MixMonitorID can be stored as a channel variable using
  the 'i' MixMonitor option and is returned upon creation if
  this option is used.
  As part of this change, if no MixMonitorID is specified in
  the manager action MixMonitorMute, Asterisk will set the mute
  flag on all MixMonitor audiohooks on the channel.  Previous
  behavior would set the flag on the first MixMonitor audiohook
  found.

- ### pbx_dundi: Add PJSIP support.
  DUNDi now supports chan_pjsip. Outgoing calls using
  PJSIP require the pjsip_outgoing_endpoint option
  to be set in dundi.conf.

- ### test.c: Fix counting of tests and add 2 new tests
  The "tests" attribute of the "testsuite" element in the
  output XML now reflects only the tests actually requested
  to be executed instead of all the tests registered.
  The "failures" attribute was added to the "testsuite"
  element.
  Also added two new unit tests that just pass and fail
  to be used for testing CI itself.

- ### cli: increase channel column width
  This change increases the display width on 'core show channels'
  amd 'core show channels verbose'
  For 'core show channels', the Channel name field is increased to
  64 characters and the Location name field is increased to 32
  characters.
  For 'core show channels verbose', the Channel name field is
  increased to 80 characters, the Context is increased to 24
  characters and the Extension is increased to 24 characters.


Upgrade Notes:
----------------------------------------


Closed Issues:
----------------------------------------

  - #193: [bug]: third-party/apply-patches doesn't sort the patch file list before applying


                               [asterisk-announce] Asterisk Release 18.18.0

The Asterisk Development Team would like to announce
the release of Asterisk 18.18.0.

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!


Change Log for Release 18.18.0
========================================

Summary:
----------------------------------------

- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize

User Notes:
----------------------------------------

- ### cel: add local optimization begin event
  The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
  by itself or in conert with the existing
  AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.

- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
  A "dialmode" option has been added which allows
  specifying, on a per-channel basis, what methods of
  subscriber dialing (pulse and/or tone) are permitted.
  Additionally, this can be changed on a channel
  at any point during a call using the CHANNEL
  function.

- ### pbx_dundi: Add PJSIP support.
  DUNDi now supports chan_pjsip. Outgoing calls using
  PJSIP require the pjsip_outgoing_endpoint option
  to be set in dundi.conf.

- ### cli: increase channel column width
  This change increases the display width on 'core show channels'
  amd 'core show channels verbose'
  For 'core show channels', the Channel name field is increased to
  64 characters and the Location name field is increased to 32
  characters.
  For 'core show channels verbose', the Channel name field is
  increased to 80 characters, the Context is increased to 24
  characters and the Extension is increased to 24 characters.

- ### app_senddtmf: Add SendFlash AMI action.
  The SendFlash AMI action now allows sending
  a hook flash event on a channel.

- ### res_http_media_cache: Introduce options and customize
  The res_http_media_cache module now attempts to load
  configuration from the res_http_media_cache.conf file.
  The following options were added:
    * timeout_secs
    * user_agent
    * follow_location
    * max_redirects
    * protocols
    * redirect_protocols
    * dns_cache_timeout_secs

- ### test.c: Fix counting of tests and add 2 new tests
  The "tests" attribute of the "testsuite" element in the
  output XML now reflects only the tests actually requested
  to be executed instead of all the tests registered.
  The "failures" attribute was added to the "testsuite"
  element.
  Also added two new unit tests that just pass and fail
  to be used for testing CI itself.

- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
  It is now possible to specify the MixMonitorID when calling
  the manager action: MixMonitorMute.  This will allow an
  individual MixMonitor instance to be muted via ID.
  The MixMonitorID can be stored as a channel variable using
  the 'i' MixMonitor option and is returned upon creation if
  this option is used.
  As part of this change, if no MixMonitorID is specified in
  the manager action MixMonitorMute, Asterisk will set the mute
  flag on all MixMonitor audiohooks on the channel.  Previous
  behavior would set the flag on the first MixMonitor audiohook
  found.

- ### bridge_builtin_features: add beep via touch variable
  Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
  Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid
  interval in seconds will result in a periodic beep being
  played to the monitored channel upon MixMontior/Monitor
  feature start.
  If an interval less than 5 seconds is specified, the interval
  will default to 5 seconds.  If the value is set to an invalid
  interval, the default of 15 seconds will be used.

- ### format_sln: add .slin as supported file extension
  format_sln now recognizes '.slin' as a valid
  file extension in addition to the existing
  '.sln' and '.raw'.


Upgrade Notes:
----------------------------------------

- ### cel: add local optimization begin event
  The existing AST_CEL_LOCAL_OPTIMIZE can continue
  to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
  can be ignored if desired.


Closed Issues:
----------------------------------------

  - #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
  - #39: [Bug]: Remove .gitreview from repository.
  - #43: [Bug]: Link to trademark policy is no longer correct
  - #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
  - #52: [improvement]: Add local optimization begin cel event

### For more details, see:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md


                            [asterisk-announce] Asterisk 18.17.1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.17.1.

The release of Asterisk 18.17.1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-30469 - res_pjsip_pubsub: Regression for
      subscription shutdowns
      (Reported by N A)
 * ASTERISK-30472 - pbx_ael: Literal usage for variables broken
      (Reported by isrl)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1

Thank you for your continued support of Asterisk!


                            [asterisk-announce] Asterisk 18.17.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.17.0.

The release of Asterisk 18.17.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-29810 - app_signal: Add channel signaling
      applications
      (Reported by N A)
 * ASTERISK-30262 - res_pjsip_session: Allow a context to be
      specified for overlap dialing
      (Reported by N A)
 * ASTERISK-30319 - Add BYE Reason support for SIP
      (Reported by Igor Goncharovsky)
 * ASTERISK-30180 - app_broadcast: Add a channel audio
      multicasting application
      (Reported by N A)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-27830 - Asterisk crashes on Invalid UTF-8 string
      (Reported by AvayaXAsterisk)
 * ASTERISK-30354 - chan_iax2: Lack of formats prior to
      receiving voice frames causes jitterbuffer to stall
      (Reported by N A)
 * ASTERISK-30162 - when chan_iax is used to relay calls, no
      ringing indication is played
      (Reported by Jaco Kroon)
 * ASTERISK-30424 - pjproject_bundled: cross-compilation broken
      when ssl autodetected
      (Reported by Nick French)
 * ASTERISK-30388 - res_phoneprov: Stale SERVER variable when
      multi-homed
      (Reported by cmaj)
 * ASTERISK-30419 - pjsip: Crash when sending NOTIFY in PJSIP 2.13
      (Reported by Ross Beer)
 * ASTERISK-30417 - Copy/Paste error in UnpauseQueueMember
      (Reported by Sean Bright)
 * ASTERISK-30406 - pbx_ael: Global variables are not expanded.
      (Reported by Sean Bright)
 * ASTERISK-29604 - ari: Segfault with lots of calls
      (Reported by Danila Evgrafov)
 * ASTERISK-30391 - res_rtp_asterisk: Issue with transcoding
      g722 after MES changes
      (Reported by George Joseph)
 * ASTERISK-30345 - loader.c: Modules that decline to load
      cannot be reloaded
      (Reported by N A)
 * ASTERISK-30379 - http: fix NULL pointer dereference while
      enable_status on TLS-only
      (Reported by Boris P. Korzun)
 * ASTERISK-30375 - res_http_media_cache: Crash when URL has no
      path component.
      (Reported by Sean Bright)
 * ASTERISK-30351 - manager: Originate variables are not added
      when setvar used in manager.conf
      (Reported by Sebastian
      Gutierrez)
 * ASTERISK-30369 - res_pjsip: Websockets from same IP shut down
      when they shouldn't be
      (Reported by Joshua C. Colp)
 * ASTERISK-30367 - pbx: Fix outdated channel snapshots with pbx_exec
      (Reported by N A)
 * ASTERISK-28767 - chan_pjsip: Caller ID not used when checking
      for extension, callerid supplement executed too late
      (Reported by Oleg)
 * ASTERISK-30350 - res_pjsip_sdp_rtp: rtp_timeout_hold is not
      used when moh_passthrough has call on hold
      (Reported by Benjamin Keith Ford)
 * ASTERISK-30240 - app voicemail odbc build error with gcc 11.1
      (Reported by Michael Bradeen)
 * ASTERISK-30100 - res_pjsip: Path is ignored on INVITE to endpoint
      (Reported by Yury Kirsanov)
 * ASTERISK-30198 - Error `Too many open files` occurs after
      about ~8000 calls when using mixmonitor
      (Reported by Julien Alie)

Improvements made in this release:
-----------------------------------
 * ASTERISK-30411 - app_read: add option to include terminating
      digit on empty, terminated strings
      (Reported by Michael Bradeen)
 * ASTERISK-30405 - app_directory: Add 's' option to skip
      channel call
      (Reported by Michael Bradeen)
 * ASTERISK-30422 - app_senddtmf: add the option for senddtmf to answer
      (Reported by Michael Bradeen)
 * ASTERISK-30325 - Upgrade Asterisk to bundled pjproject 2.13
      (Reported by Stanislav Abramenkov)
 * ASTERISK-30404 - app_directory: Add reading directory
      configuration from custom file
      (Reported by Michael Bradeen)
 * ASTERISK-29913 - func_json: Adds multi-level and array
      parsing to JSON_DECODE
      (Reported by N A)
 * ASTERISK-30353 - func_frame_trace: Print text for text frames
      (Reported by N A)
 * ASTERISK-30361 - json.h: Add missing
      ast_json_object_real_get
      (Reported by N A)
 * ASTERISK-30280 - Create capability to assign a Media
      Experience Score to RTP streams
      (Reported by George Joseph)
 * ASTERISK-30332 - func_callerid: Warn if invalid redirecting
      reason provided
      (Reported by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0

Thank you for your continued support of Asterisk!


               [asterisk-announce] Asterisk 16.29.1, 18.15.1, 19.7.1, 20.0.1 Now Available

The Asterisk Development Team would like to announce the release of
Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1.

The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues reported by the
community and would have not been possible without your participation.Thank you!

The following issue is resolved in this release:

Bugs fixed in this release:
-----------------------

[ASTERISK-30103 <https://issues.asterisk.org/jira/browse/ASTERISK-30103>]
chan_ooh323 vulnerability in calling/called party IE (Reported By: Michael
Bradeen)

[ASTERISK-30176 <https://issues.asterisk.org/jira/browse/ASTERISK-30176>]
GetConfig can read files outside of Asterisk (Reported By: shawty)

[ASTERISK-30244 <https://issues.asterisk.org/jira/browse/ASTERISK-30244>]
Occasional crash when TCP/TLS connection terminated and subscription
persistence is removed (Reported By: nappsoft)

[ASTERISK-30338 <https://issues.asterisk.org/jira/browse/ASTERISK-30338>]
Backport 2.13 security fixes from pjproject


                            [asterisk-announce] Asterisk 18.15.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.15.0.

The release of Asterisk 18.15.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-30037 - Add test support to calling external processes
      (Reported by Philip Prindeville)
 * ASTERISK-30161 - locks: add AMI event for deadlock
      (Reported by N A)
 * ASTERISK-30211 - app_confbridge: Add end_marked_any option
      (Reported by N A)
 * ASTERISK-30186 - res_pjsip: Add support for reloading TLS
      certificate and key information
      (Reported by Joshua C.  Colp)
 * ASTERISK-29899 - features: Add advanced transfer initiation options
      (Reported by N A)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-30235 - res_crypto and tests:  Memory issues and and
      uninitialized variable error
      (Reported by George Joseph)
 * ASTERISK-30234 - res_geolocation: ...may be used
      uninitialized error in geoloc_config.c
      (Reported by George Joseph)
 * ASTERISK-30215 - Inbound SIP INVITE with Geo Location causing
      a Segmentation Fault
      (Reported by Dan Cropp)
 * ASTERISK-30135 - [res_musiconhold] Allows the moh only for
      the answered call
      (Reported by sungtae kim)
 * ASTERISK-26894 - pjsip should support tel uri scheme
      (Reported by Gergely D½½ms½½di)
 * ASTERISK-30210 - func_frame_trace: Channel masquerade
      triggers assertion
      (Reported by N A)
 * ASTERISK-30190 - res_geolocation:  GEOLOC_PROFILE isn't
      returning correct values on incoming channel
      (Reported by George Joseph)
 * ASTERISK-29185 - chan_pjsip: Endpoint: allow = all is broken.
      (Reported by Alexander Traud)
 * ASTERISK-30192 - res_tonedetect: fix typo for frametype
      (Reported by N A)
 * ASTERISK-29453 - alembic: incoming_call_offer_pref and
      outgoing_call_offer_pref missing in "ps_endpoints" table
      (Reported by Daniel Th½½men)
 * ASTERISK-26826 - testsuite: Add support for Python 3
      (Reported by Joshua C. Colp)
 * ASTERISK-30167 - res_geolocation:  Refactor for issues found
      by users
      (Reported by George Joseph)
 * ASTERISK-28422 - Memory Leak in Confbridge menu
      (Reported by Ted G)
 * ASTERISK-29917 - ami: FilterList action doesn't exist
      (Reported by N A)
 * ASTERISK-30018 - app_meetme: MeetmeList AMI event not documented
      (Reported by Michael Cargile)
 * ASTERISK-30020 - ConfbridgeListRooms Event Not Documented
      (Reported by Michael Cargile)
 * ASTERISK-30151 - Documentation doesn't include info about
      "field", a 3rd required parameter.
      (Reported by Chris Young)

Improvements made in this release:
-----------------------------------
 * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
      scope trace debugs to DEBUG level
      (Reported by N A)
 * ASTERISK-30178 - extend user_eq_phone behavior to local uri's
      (Reported by Michael Bradeen)
 * ASTERISK-30046 - Reimplement res/res_crypto.c internals with
      EVP_PKEY interface to Openssl API's
      (Reported by Philip Prindeville)
 * ASTERISK-30045 - Add test coverage to res/res_crypto.c functionality
      (Reported by Philip Prindeville)
 * ASTERISK-30185 - res_geolocation: Allow location parameters
      to be specified in profiles
      (Reported by George Joseph)
 * ASTERISK-30177 - res_geolocation:  Add option to suppress
      empty elements
      (Reported by George Joseph)
 * ASTERISK-30182 - res_geolocation: Add built-in profiles to
      use in fully dynamic configurations
      (Reported by George Joseph)
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
      the lists
      (Reported by Alexei Gradinari)
 * ASTERISK-30163 - general: fix minor formatting issues
      (Reported by N A)
 * ASTERISK-30164 - chan_iax2: Add missing option documentation
      (Reported by N A)
 * ASTERISK-30153 - logger: Improve log levels
      (Reported by N A)
 * ASTERISK-30160 - cdr.conf: Remove obsolete app_mysql reference
      (Reported by N A)
 * ASTERISK-30159 - general: Remove obsolete SVN references
      (Reported by N A)


                            [asterisk-announce] Asterisk 18.14.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.14.0.

The release of Asterisk 18.14.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Improvements made in this release:
-----------------------------------
 * ASTERISK-30128 - Create PJSIP interface module for Geolocation
      (Reported by George Joseph)
 * ASTERISK-30127 - Create core Geolocation capability for Asterisk
      (Reported by George Joseph)
 * ASTERISK-30089 - general: fix typos
      (Reported by N A)
 * ASTERISK-30050 - Upgrade Asterisk to bundled pjproject 2.12.1
      (Reported by Stanislav Abramenkov)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-30167 - res_geolocation:  Refactor for issues found
      by users
      (Reported by George Joseph)
 * ASTERISK-29966 - pbx_variables: ast_str_strlen can be wrong
      (Reported by N A)
 * ASTERISK-29905 - OSX: bininstall launchd issue on
      cross-platfrom build
      (Reported by Sergey V. Lobanov)
 * ASTERISK-30137 - manager: Global disabled event filtered is incomplete
      (Reported by N A)
 * ASTERISK-30109 - res_pjsip: no contact-status AMI event on
      register of prune-on-boot contact that uses the same URI as
      before Asterisk restart
      (Reported by Michael Neuhauser)
 * ASTERISK-29991 - chan_dahdi, callerid: Caller ID does not
      honor presentation
      (Reported by N A)
 * ASTERISK-30126 - Spelling mistake in
      configs/samples/queues.conf.sample
      (Reported by Sam Banks)
 * ASTERISK-30029 - build: Git security vulnerability fix is sad
      with our accessing git as root during "make install"
      (Reported by Joshua C. Colp)
 * ASTERISK-29907 - res_pjsip, app_confbridge: Video call
      through ConfBridge with normal endpoints causes infinite
      loop/crash
      (Reported by N A)
 * ASTERISK-30138 - Compile failure in
      res_geolocation/geoloc_eprofile.c when optimization is enabled
      (Reported by George Joseph)
 * ASTERISK-30096 -  cel_odbc: Column type 9 (field
      'cdr:cel:eventtime') is unsupported at this time
      (Reported by Morvai Szabolcs)
 * ASTERISK-30083 - chan_iax2: Optional dependency on
      openssl/res_crypto is now mandatory
      (Reported by Dmitry Melekhov)
 * ASTERISK-30099 - test_aeap_transport: transport_connect_fail
      sporadically causes failure
      (Reported by Kevin Harwell)
 * ASTERISK-30123 - features: Update automixmon documentation to
      reflect reality
      (Reported by Trevor Peirce)
 * ASTERISK-30117 - pbx_lua: Remove compiler warnings
      (Reported by Boris P. Korzun)
 * ASTERISK-30101 - res_prometheus: Optional load
      res_pjsip_outbound_registration.so
      (Reported by Boris P.  Korzun)
 * ASTERISK-29989 - app_dial, chan_dahdi: DIALSTATUS is
      inconsistent for busy
      (Reported by N A)
 * ASTERISK-30001 - db: Removing nonexistent entries shows
      "Database entry removed"
      (Reported by N A)
 * ASTERISK-30115 - app_dial: Allow hook flashes to propogate on
      outbound dials
      (Reported by N A)
 * ASTERISK-30106 - res_calendar_icalendar: Microsoft online ICS
      calendars no longer work
      (Reported by N A)
 * ASTERISK-29822 - cli: Typing \? freezes the CLI permanently
      with remote console
      (Reported by N A)
 * ASTERISK-30072 - res_pjsip: allow TLS verification of
      wildcard cert-bearing servers
      (Reported by Kevin Harwell)
 * ASTERISK-30075 - say: Abort if channel hangs up during playback
      (Reported by N A)

New Features made in this release:
-----------------------------------
 * ASTERISK-30136 - db: Add AMI action to retrieve all keys
      beginning with a prefix
      (Reported by N A)
 * ASTERISK-30000 - chan_dahdi: Add POLARITY function
      (Reported by N A)
 * ASTERISK-30062 - cli: Add CLI command to execute a dialplan app
      (Reported by N A)
 * ASTERISK-29999 - pjsip: Get information from 200 OK INVITE
      reply headers
      (Reported by Jos½½ Lopes)
 * ASTERISK-30061 - pbx: Add pbx helper application
      (Reported by N A)


                            [asterisk-announce] Asterisk 18.13.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.13.0.

The release of Asterisk 18.13.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Improvements made in this release:
-----------------------------------
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to the lists
      (Reported by Alexei Gradinari)
 * ASTERISK-29891 - [patch] provide a display name for RLS subscriptions
      (Reported by Alexei Gradinari)
 * ASTERISK-30090 - xmldocs: Use example tags for examples
      (Reported by N A)
 * ASTERISK-30086 - res_parking: Warn when invalid parking space requested
      (Reported by N A)
 * ASTERISK-30058 - Evaluate dialplan functions and variables in agi exec
      (Reported by Shloime Rosenblum)
 * ASTERISK-30027 - ari: expose channel driver's unique id (i.e.
      Call-ID for chan_sip/chan_pjsip) in ARI channel resource
      (Reported by Moritz Fain)
 * ASTERISK-29845 - res_pjsip_outbound_registration: Show time
      remaining until registration lapses
      (Reported by N A)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-30097 - console: Recent documentation changes for
      connecting to remote console are inconsistent
      (Reported by Matthias Hensler)
 * ASTERISK-30043 - Wrong party is disconnected when
      hook-flashing on 3-way bridge
      (Reported by Josh Alberts)
 * ASTERISK-29603 - res_pjsip: UPDATE/re-INVITE not sent when
      "timers=always" is specified in pjsip.conf
      (Reported by Ray Crumrine)
 * ASTERISK-30092 - DateTime application: wrong inflection for
      one o'clock in German
      (Reported by Christof Efkemann)
 * ASTERISK-30064 - pbx: iax2 switch causes crash due to
      deadlock and assertion
      (Reported by N A)
 * ASTERISK-29981 - res_calendar: Asterisk crashes when
      starting, and will not run
      (Reported by N A)
 * ASTERISK-30039 - cli: Targeted debug on startup deadlocks and
      creates unstable system
      (Reported by N A)
 * ASTERISK-30051 - res_pjsip: No video after un-hold with
      moh_passthrough=yes
      (Reported by Maximilian Fridrich)
 * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
      in PJSIP NOTIFY event: dialog  XML body
      (Reported by Marco Paland)
 * ASTERISK-30059 - menuselect: libxml include fails under Gentoo
      (Reported by waltermoeller)
 * ASTERISK-30060 - loader: format warnings in dev mode
      (Reported by N A)
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
      reused for outgoing requests
      (Reported by LA)
 * ASTERISK-30042 - res_pjsip_transport_websocket: Registration
      over websocket returns a rewritten contact
      (Reported by Thomas Guebels)
 * ASTERISK-29993 - chan_dahdi: Operator control option borks
      both lines involved on callee disconnect
      (Reported by N A)
 * ASTERISK-30044 - GCC 12 issues
      (Reported by George Joseph)

New Features made in this release:
-----------------------------------
 * ASTERISK-30063 - app_voicemail: Add option to prevent
      deletion of messages
      (Reported by N A)
 * ASTERISK-29965 - res_pjsip_outbound_registration: Make max
      registration delay configurable
      (Reported by N A)
 * ASTERISK-30087 - res_parking: Add music on hold override option
      (Reported by N A)
 * ASTERISK-30036 - app_confbridge: Add CONFBRIDGE_CHANNELS function
      (Reported by N A)

Thank you for your continued support of Asterisk!


                            [asterisk-announce] Asterisk 18.12.1 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.12.1.

The release of Asterisk 18.12.1 resolves an issue reported by the
community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
      reused for outgoing requests
      (Reported by LA)


                            [asterisk-announce] Asterisk 18.12.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 18.12.0.

The release of Asterisk 18.12.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Security bugs fixed in this release:
-----------------------------------
 * ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities
      (Reported by Clint Ruoho)
 * ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a terminating \
      (Reported by Leandro Dardini)
 * ASTERISK-29872 - res_stir_shaken: Resource exhaustion with large files
      (Reported by Benjamin Keith Ford)

New Features made in this release:
-----------------------------------
 * ASTERISK-29931 - Option to allow a user to not hear the join
      sound on enter but everyone else can
      (Reported by Michael Cargile)
 * ASTERISK-29968 - func_db: Add a function to return
      cardinality of keys at prefix
      (Reported by N A)
 * ASTERISK-29486 - Hint-like extension value lookup function
      without device state
      (Reported by N A)
 * ASTERISK-29941 - chan_pjsip: Add ability to send flash events
      (Reported by N A)
 * ASTERISK-29820 - cli: Add command to evaluate a function
      (Reported by N A)
 * ASTERISK-29876 - app_queue: Add music on hold option
      (Reported by N A)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-29655 - res_pjsip_session: No video to caller if no
      camera available
      (Reported by Michael Auracher)
 * ASTERISK-29638 - res_pjsip_session: No video after early media
      (Reported by Michael Auracher)
 * ASTERISK-28518 - chan_dahdi: Caller ID FSK Erroneously Sent
      when Picking Up Dahdi Call On Hold
      (Reported by Josh Alberts)
 * ASTERISK-29990 - chan_dahdi: adding ring cadences is not
      idempotent on dahdi restart
      (Reported by N A)
 * ASTERISK-30007 - chan_iax2: Prevent crashes due to attempted
      encryption with missing secrets
      (Reported by N A)
 * ASTERISK-29728 - menuselect: Disabled by default modules that
      are enabled are always recompiled
      (Reported by N A)
 * ASTERISK-30002 - app_meetme: Don't erroneously set global
      variables when channel is NULL
      (Reported by N A)
 * ASTERISK-29994 - chan_dahdi: Round robin array size is too
      small for max number of groups
      (Reported by N A)
 * ASTERISK-22246 - Asterisk's "T" flag is ignored when used
      with "r" or "R" flags. (documentation bug)
      (Reported by Rusty Newton)
 * ASTERISK-26582 - Asterisk seems to ignore the "n" parameter
      for "disable console colorization"
      (Reported by Sebastian Gutierrez)
 * ASTERISK-29843 - Session timers get removed on UPDATE
      (Reported by Mark Petersen)
 * ASTERISK-29943 - file.c: seeking to negative file offset is
      not prevented
      (Reported by N A)
 * ASTERISK-29955 - chan_sip: SIP route header is missing on UPDATE
      (Reported by Mark Petersen)
 * ASTERISK-29842 - Do not change 180 Ringing to 183 Progress
      even if early_media already enabled
      (Reported by Mark Petersen)
 * ASTERISK-29948 - iostream: Infinite TCP timeout writing data
      (Reported by N A)
 * ASTERISK-29253 - Incorrect bridging on transfer
      (Reported by Yury Kirsanov)
 * ASTERISK-30006 - res_pjsip: UDP transport does not work when
      async_operations is greater than 1
      (Reported by Ross Beer)
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
      functionality not enabled
      (Reported by Claude Diderich)
 * ASTERISK-30021 - ast_variable_list_replace_variable uses
      variable with new keyword
      (Reported by Jasper Hafkenscheid)
 * ASTERISK-30023 - cdr_adaptive_odbc: does not support DATETIME
      database columns
      (Reported by Gregory Massel)
 * ASTERISK-30015 - pjsip / WebRTC: Chrome creating large number
      of SDP attributes
      (Reported by Josh Hogan)
 * ASTERISK-26689 - res_pjsip_sdp_rtp: 183 Session in Progress.
      Disconnecting channel for lack of RTP activity
      (Reported by Dmitriy Serov)
 * ASTERISK-29929 - res_pjsip_sdp_rtp: Disconnecting channel for
      lack of RTP activity in one way sessions
      (Reported by Boris P. Korzun)
 * ASTERISK-29411 - Crash in pjsip_msg_find_hdr_by_name
      (Reported by LA)
 * ASTERISK-29535 - Segmentation fault in libasteriskpj.so.2
      (Reported by Daniel Bonazzi)
 * ASTERISK-26719 - pbx: Only up to 127 includes in a dialplan
      context (AST_PBX_MAX_STACK - 1)
      (Reported by Tzafrir Cohen)
 * ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
      wget isn't available
      (Reported by Stefan Ruijsenaars)
 * ASTERISK-29988 - REGRESSION: The build process is requiring
      xmllint or xmlstarlet ro be installed when it shouldn't
      (Reported by George Joseph)
 * ASTERISK-29895 - chan_iax2: Fix misaligned spacing in iax2
      show netstats printout
      (Reported by N A)
 * ASTERISK-29939 - agi: Fix xmldoc bug with set music
      (Reported by N A)
 * ASTERISK-28891 - documentation: AGICommand_set+music
      documentation arguments displayed incorreclty
      (Reported by Jonathan Harris)
 * ASTERISK-29048 - chan_iax2: "iax2 show registry" shows host
      for perceived
      (Reported by David Herselman)
 * ASTERISK-29674 - Adjust for 64bit time_t
      (Reported by Andre Heider)
 * ASTERISK-29961 - RLS: domain part of 'uri' list attribute
      mismatch with SUBSCRIBE request
      (Reported by Alexei Gradinari)
 * ASTERISK-29928 - logging messages truncated when using MUSL runtime
      (Reported by Philip Prindeville)
 * ASTERISK-29960 - ari: Retrieving stored recording can returns
      wrong file
      (Reported by Arix)
 * ASTERISK-29950 - SayNumber can handle '01' to '07', but not
      '08' or '09'
      (Reported by Jim Van Meggelen)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24827 - Missing documentation for chan_dahdi dial
      string ring cadences
      (Reported by Scott Griepentrog)
 * ASTERISK-29940 - general: Add since tags to xmldocs
      (Reported by N A)
 * ASTERISK-29726 - Add Asterisk External Application Protocol
      (AEAP) implementation
      (Reported by Kevin Harwell)
 * ASTERISK-29951 - app_mf, app_sf: Return -1 on hangup
      (Reported by N A)
 * ASTERISK-29954 - app_meetme: Emit warning if conference not found
      (Reported by N A)
 * ASTERISK-29351 - Qualify pjproject 2.12 for Asterisk
      (Reported by George Joseph)
 * ASTERISK-29976 - Should Readme include information about
      install_prereq script?
      (Reported by Marcel Wagner)
 * ASTERISK-29970 - Use pkg-config to find libxml2 headers and libraries
      (Reported by Hugh McMaster)
 * ASTERISK-29980 - build: External binary modules don't use https
      (Reported by INVADE International Ltd.)
 * ASTERISK-25716 - Documentation: Document explanations and
      examples for possible values of DIALSTATUS
      (Reported by Rusty Newton)
 * ASTERISK-29967 - pbx_builtins: Add missing documentation
      (Reported by N A)


                            [asterisk-announce] Asterisk 18.11.3 Now Available

   Asterisk Development Team asteriskteam at digium.com
   Tue Apr 26 12:09:50 CDT 2022

The Asterisk Development Team would like to announce the release of Asterisk 18.11.3.

The release of Asterisk 18.11.3 resolves an issue reported by the
community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
      functionality not enabled
      (Reported by Claude Diderich)


      [asterisk-announce] Asterisk 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14 Now Available (Security)

The Asterisk Development Team would like to announce security releases for
Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases are
released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.

The following security vulnerabilities were resolved in these versions:

* AST-2022-001: res_stir_shaken: resource exhaustion with large files
  When using STIR/SHAKEN, it½½½s possible to download files that are not
  certificates. These files could be much larger than what you would expect to
  download.

* AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header
  When using STIR/SHAKEN, it½½½s possible to send arbitrary requests like GET to
  interfaces such as localhost using the Identity header.

* AST-2022-003: func_odbc: Possible SQL Injection
  Some databases can use backslashes to escape certain characters, such as
  backticks. If input is provided to func_odbc which includes backslashes it is
  possible for func_odbc to construct a broken SQL query and the SQL query to
  fail.


                            [asterisk-announce] Asterisk 18.11.1 Now Available

   Asterisk Development Team asteriskteam at digium.com
   Tue Mar 29 19:15:43 CDT 2022

The Asterisk Development Team would like to announce the release of Asterisk 18.11.1.

The release of Asterisk 18.11.1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
      wget isn't available
      (Reported by Stefan Ruijsenaars)
 * ASTERISK-29988 - REGRESSION: The build process is requiring
      xmllint or xmlstarlet ro be installed when it shouldn't
      (Reported by George Joseph)


                            [asterisk-announce] Asterisk 18.11.0 Now Available

   Asterisk Development Team asteriskteam at digium.com
   Thu Mar 24 09:06:03 CDT 2022

The Asterisk Development Team would like to announce the release of Asterisk 18.11.0.

The release of Asterisk 18.11.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Security bugs fixed in this release:
-----------------------------------
 * ASTERISK-29945 - pjproject: Security fixes for things
      (Reported by Kevin Harwell)

New Features made in this release:
-----------------------------------
 * ASTERISK-29853 - ami: Allow events to be globally disabled
      (Reported by N A)
 * ASTERISK-29840 - func_channel: Add LASTCONTEXT and LASTEXTEN fields
      (Reported by N A)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-29924 - res_config_pgsql: omit "unsupported column
      type 'text'" error
      (Reported by Boris P. Korzun)
 * ASTERISK-29923 - docs, LICENSE: pbx.digium.com no longer exists
      (Reported by N A)
 * ASTERISK-29904 - RLS: Batched Notifications stop working
      (Reported by Alexei Gradinari)
 * ASTERISK-29365 - taskprocessor: Can cause assert at shutdown
      (Reported by Joshua C. Colp)
 * ASTERISK-29873 - [patch] Queue Realtime load
      (Reported by Alexei Gradinari)
 * ASTERISK-18416 - [patch] Realtime queue agents unavailable
      via AMI before a call event.
      (Reported by kwk)
 * ASTERISK-27597 - AMI Queuestatus not working (with realtime queue)
      (Reported by cagdas kopuz)
 * ASTERISK-29871 - res_prometheus: Failure to load causes FRACKs
      (Reported by Mark Petersen)
 * ASTERISK-29886 - Asterisk AMI sends not-valid XML
      (Reported by Napadailo Yaroslav)

Improvements made in this release:
-----------------------------------
 * ASTERISK-29909 - app_queue: Add support for withdrawing a call
      (Reported by Kfir Itzhak)
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
      the lists
      (Reported by Alexei Gradinari)
 * ASTERISK-29353 - Qualify jansson 2.14 for asterisk
      (Reported by George Joseph)
 * ASTERISK-29897 - channels: Increase core debug levels for
      chatty debugs
      (Reported by N A)
 * ASTERISK-29896 - xmldocs: Add since tag
      (Reported by N A)
 * ASTERISK-29861 - asterisk.h: add macro for curl user agent
      (Reported by N A)
 * ASTERISK-29809 - curl, stir_shaken: refactor curl code
      (Reported by N A)
 * ASTERISK-29920 - app_voicemail: Warn if trying to manage
      nonexistent mailbox
      (Reported by N A)
 * ASTERISK-29925 - func_db: Warn about malformed key names
      (Reported by N A)
 * ASTERISK-29891 - [patch] provide a display name for RLS subscriptions
      (Reported by Alexei Gradinari)
 * ASTERISK-29866 - cli: add core dump information to core show settings
      (Reported by N A)
 * ASTERISK-29898 - documentation: Add default attributes to documentation
      (Reported by N A)
 * ASTERISK-29900 - app_mp3: Document and warn about https incompatibility
      (Reported by N A)
 * ASTERISK-29877 - app_mf: Allow reading a maximum number of digits
      (Reported by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0


      [asterisk-announce] Asterisk 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13 Now Available (Security)

   Asterisk Development Team asteriskteam at digium.com
   Fri Mar 4 14:01:59 CST 2022

The Asterisk Development Team would like to announce security releases for
Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases are
released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.

The following security vulnerabilities were resolved in these versions:

* AST-2022-004: pjproject: integer underflow on STUN message
  The header length on incoming STUN messages that contain an ERROR-CODE
  attribute is not properly checked. This can result in an integer underflow.
  Note, this requires ICE or WebRTC support to be in use with a malicious remote
  party.

* AST-2022-005: pjproject: undefined behavior after freeing a dialog set
  When acting as a UAC, and when placing an outgoing call to a target that then
  forks Asterisk may experience undefined behavior (crashes, hangs, etc½½½)
  after a dialog set is prematurely freed.

* AST-2022-006: pjproject: unconstrained malformed multipart SIP message
  If an incoming SIP message contains a malformed multi-part body an out of
  bounds read access may occur, which can result in undefined behavior. Note,
  it½½½s currently uncertain if there is any externally exploitable vector
  within Asterisk for this issue, but providing this as a security issue out of
  caution.

The security advisories are available at:

https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
https://downloads.asterisk.org/pub/security/AST-2022-006.pdf


                            [asterisk-announce] Asterisk 18.10.0 Now Available

   Asterisk Development Team asteriskteam at digium.com
   Thu Feb 10 06:58:17 CST 2022

The Asterisk Development Team would like to announce the release of Asterisk 18.10.0.

The release of Asterisk 18.10.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-29808 - cdr: allow disabling CDR by default
      (Reported by N A)
 * ASTERISK-29830 - ami: Add AMI event for Wink
      (Reported by N A)
 * ASTERISK-29802 - app_sf: Add full tech-agnostic SF support
      (Reported by N A)
 * ASTERISK-29759 - app_sendtext: Add ReceiveText application
      (Reported by N A)
 * ASTERISK-29706 - func_json: Add JSON parsing function
      (Reported by N A)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-29888 - res_pjsip_outbound_authenticator_digest:
      ABRT attempting to clean up auth_sess
      (Reported by George Joseph)
 * ASTERISK-29857 - res_tonedetect: fix logic errors in code
      (Reported by N A)
 * ASTERISK-29854 - func_frame_drop: fix buffer usage typo
      (Reported by N A)
 * ASTERISK-29869 - rtp sequence number can skip after DTMF
      under certain bridges
      (Reported by Torrey Searle)
 * ASTERISK-29817 - gethostbyname_r is misdetected on NetBSD and
      causes a build failure
      (Reported by Micha½½ G½½rny)
 * ASTERISK-29698 - Segfault if sorcery object_lifetime_maximum
      and qualify_frequency the same value
      (Reported by Alexei Gradinari)
 * ASTERISK-29851 - rdtsc is not enabled (stubbed out) on NetBSD
      (Reported by Micha½½ G½½rny)
 * ASTERISK-29852 - make_version uses GNU-ism that break
      git-svn-id parsing on NetBSD
      (Reported by Micha½½ G½½rny)
 * ASTERISK-29850 - ast_get_tid() not implemented for NetBSD
      (Reported by Micha½½ G½½rny)
 * ASTERISK-29818 - Build failure on NetBSD due to hmac function collision
      (Reported by Micha½½ G½½rny)
 * ASTERISK-29856 - res_rtp_asterisk: Invalid comparison creates
      unreachable code
      (Reported by N A)
 * ASTERISK-29867 - configure fails if libsrtp dev files are not installed
      (Reported by Sean Bright)
 * ASTERISK-29813 - res_pjsip_session doesn't support multipart
      message bodies
      (Reported by George Joseph)
 * ASTERISK-29858 - Regression:  Using external pjproject not
      working after "hack" commit
      (Reported by George Joseph)
 * ASTERISK-29859 - VoiceMailMain() fails when encountering
      non-numeric CALLERID(num)
      (Reported by Mark Murawski)
 * ASTERISK-29847 - pbx_variables: ASTSBINDIR is missing
      (Reported by N A)
 * ASTERISK-29824 - It's hard to make changes to bundled pjproject
      (Reported by George Joseph)
 * ASTERISK-29695 - SAY.CONF wrong logic when converting 24hour
      time to say 12 hour am/pm
      (Reported by Vincent Dubois)
 * ASTERISK-29664 - PJSIP processing token with % incorrectly
      (Reported by Dan Cropp)
 * ASTERISK-29827 - Support for Nordic language syntax in Queues
      (Reported by Mark Petersen)
 * ASTERISK-29515 - app_queue: QueueSummary and QueueStatus
      events don't exist in documentation
      (Reported by Luke Escude)
 * ASTERISK-29746 - tcptls.c: TCP client connect fails due to interrupt
      (Reported by Kevin Harwell)
 * ASTERISK-29806 - app_queue: extension state incorrect
      (Reported by Steve Davies)
 * ASTERISK-29816 - SAY_DTMF_INTERRUPT channel variable is not honored
      (Reported by Sean Bright)
 * ASTERISK-28863 - The ast_rtp_codecs_payloads functions don't
      preserve order
      (Reported by George Joseph)
 * ASTERISK-29320 - res_pjsip_sdp_rtp: Codec preference order of
      remote is not correct on unhold
      (Reported by Ross Beer)
 * ASTERISK-29821 - Deadlock in bridge_channel_internal_join()
      on local channels.
      (Reported by Krzysztof Trempala)
 * ASTERISK-29722 - test_timezone_watch breaks during DST to ST transition
      (Reported by Josh Soref)
 * ASTERISK-29804 - bundled_pjproject: sip_inv is missing
      multipart support in some cases
      (Reported by George Joseph)
 * ASTERISK-29794 - ast_coredumper does not delete results when
      requested and a specific output dir is set
      (Reported by Frederic Van Espen)
 * ASTERISK-29803 - pbx_variables: cp4 variables is used uninitialized
      (Reported by N A)
 * ASTERISK-29766 - pbx_variables: MSet truncates sets after 24 variables
      (Reported by N A)
 * ASTERISK-29772 - chan_sip: ${CHANNEL(ruri)} in Dial/Queue
      b(test,s,1) cause a coredump
      (Reported by Mark Petersen)
 * ASTERISK-29790 - xmldoc: Dump invalid to XML DTD: XSLT
      (Reported by Alexander Traud)
 * ASTERISK-29791 - xmldoc: Dump invalid to XML DTD: ACO Matchfield
      (Reported by Alexander Traud)
 * ASTERISK-26991 - documentation: Doxygen site is no longer
      being updated
      (Reported by Joshua C. Colp)
 * ASTERISK-20259 - [patch] Update Doxygen Configuration for
      make progdocs
      (Reported by Andrew Latham)
 * ASTERISK-29785 - res_pjsip_sdp_rtp: Warns on every offered
      crypto suite
      (Reported by Alexander Traud)
 * ASTERISK-27406 - Infinite loop when out of ports and rtpstart
      value is odd
      (Reported by Thomas Guebels)
 * ASTERISK-28053 - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
      (Reported by Simone Lazzaris)
 * ASTERISK-29761 - res: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29763 - main: Fix for Doxygen
      (Reported by Alexander Traud)

Improvements made in this release:
-----------------------------------
 * ASTERISK-29832 - Enable pickup on channel after having
      received 183 Progress
      (Reported by Mark Petersen)
 * ASTERISK-28890 - res_pjsip_sdp_rtp: Keepalive not supported
      for video streams
      (Reported by Luke Escude)
 * ASTERISK-29831 - Queue don't play "thank-you" when here is no
      hold time announcements
      (Reported by Mark Petersen)
 * ASTERISK-29855 - frame.h: fix CNG documentation typo
      (Reported by N A)
 * ASTERISK-29848 - documentation: Document special system and
      channel variables
      (Reported by N A)
 * ASTERISK-29819 - utils.c: Remove all usages of
      ast_gethostbyname()
      (Reported by Sean Bright)
 * ASTERISK-29815 - dsp: Define magic number as macro
      (Reported by N A)
 * ASTERISK-29807 - cli: add module refresh command
      (Reported by N A)
 * ASTERISK-29829 - app_mp3: Throw warning if attempting to play
      a nonexistent stream
      (Reported by N A)
 * ASTERISK-24427 - Documentation is missing for a few AMI
      Events - Including CDR and events triggered after the
      QueueStatus action
      (Reported by Dafi Ni)
 * ASTERISK-29795 - DIALEDPEERNUMBER not set on destination
      channel for Queue calls
      (Reported by Mark Petersen)
 * ASTERISK-29801 - app.c: Throw warnings for nonexistent options
      (Reported by N A)
 * ASTERISK-29797 - Support for Danish language syntax in VM
      (Reported by Mark Petersen)
 * ASTERISK-29800 - strings: Fix misusage in comment examples
      (Reported by N A)
 * ASTERISK-29758 - configs: Minor updates to sample configs
      (Reported by N A)
 * ASTERISK-29745 - pbx: Add public API for more elegant
      variable substitution with extensions
      (Reported by N A)
 * ASTERISK-29729 - Incompatibility with newer spandsp releases (3.0.0+)
      (Reported by Dustin Marquess)
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.155 2023/11/14 18:45:28 nia Exp $
d7 1
d9 1
a9 1
DISTNAME=	asterisk-18.21.0
d51 1
d165 1
a165 1
PJPROJ_VERSION=	2.13.1
d173 8
@


1.155
log
@asterisk*: Attempt to fix PLIST on SunOS
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.154 2023/11/14 14:01:31 wiz Exp $
d8 1
a8 1
DISTNAME=	asterisk-18.9.0
a9 1
PKGREVISION=	14
d163 1
a163 1
PJPROJ_VERSION=	2.10
d182 1
a182 1
SUBST_FILES.configs+=	configs/samples/sla.conf.sample
d203 6
a208 6
# XXX gross hack, gethostbyname_r on NETBSD is for internal use only
SUBST_CLASSES.NetBSD+=		gethostbyname_r
SUBST_STAGE.gethostbyname_r=	post-configure
SUBST_FILES.gethostbyname_r=	include/asterisk/autoconfig.h
SUBST_SED.gethostbyname_r=	-e "s|^\#define HAVE_GETHOSTBYNAME_R_5 1|\#undef HAVE_GETHOSTBYNAME_R_5|"
SUBST_SED.gethostbyname_r+=	-e "s|^\#define HAVE_GETHOSTBYNAME_R_6 1|\#undef HAVE_GETHOSTBYNAME_R_6|"
d239 2
a240 2
.  for f in acl.conf adsi.conf agents.conf alarmreceiver.conf alsa.conf	\
	amd.conf app_mysql.conf app_skel.conf ari.conf			\
d252 7
a258 6
	hep.conf http.conf iax.conf iaxprov.conf indications.conf	\
	logger.conf manager.conf meetme.conf mgcp.conf minivm.conf	\
	misdn.conf modules.conf	motif.conf musiconhold.conf muted.conf	\
	ooh323.conf osp.conf oss.conf phone.conf phoneprov.conf		\
	pjproject.conf pjsip.conf pjsip_notify.conf pjsip_wizard.conf	\
	queuerules.conf queues.conf res_config_mysql.conf		\
d260 6
a265 5
	res_corosync.conf res_curl.conf	 res_fax.conf res_ldap.conf	\
	res_odbc.conf res_parking.conf res_pgsql.conf res_pktccops.conf	\
	res_snmp.conf res_stun_monitor.conf resolver_unbound.conf	\
	rtp.conf say.conf sip.conf sip_notify.conf skinny.conf sla.conf	\
	smdi.conf sorcery.conf ss7.timers stasis.conf statsd.conf	\
a284 1
	${INSTALL_DATA} ${WRKSRC}/CHANGES ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
a286 1
	${INSTALL_DATA} ${WRKSRC}/ChangeLog ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
d288 1
d290 1
a290 2
	${INSTALL_DATA} ${WRKSRC}/README-SERIOUSLY.bestpractices.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/UPGRADE.txt ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
d292 9
d311 1
@


1.154
log
@*: recursive bump for cairo dependency changes
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.153 2023/11/12 13:20:51 wiz Exp $
d142 1
a142 1
PLIST_VARS+=	timerfd
d144 3
a146 1
PLIST.timerfd=	yes
@


1.153
log
@*: revebump for new brotli option for freetype2

Addresses PR 57693
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.152 2023/11/08 13:18:19 wiz Exp $
d10 1
a10 1
PKGREVISION=	13
@


1.152
log
@*: recursive bump for icu 74.1
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.151 2023/10/24 22:08:24 wiz Exp $
d10 1
a10 1
PKGREVISION=	12
@


1.151
log
@*: bump for openssl 3
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.150 2023/10/21 17:09:52 gdt Exp $
d10 1
a10 1
PKGREVISION=	11
@


1.150
log
@recursive revbump for tiff update
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.149 2023/08/14 05:23:57 wiz Exp $
d10 1
a10 1
PKGREVISION=	10
@


1.149
log
@*: recursive bump for Python 3.11 as new default
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.148 2023/04/19 08:09:21 adam Exp $
d10 1
a10 1
PKGREVISION=	9
@


1.148
log
@revbump after textproc/icu update
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.147 2023/01/29 21:15:40 ryoon Exp $
d10 1
a10 1
PKGREVISION=	8
@


1.147
log
@*: Recursive revbup from graphics/freetype2
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.146 2023/01/03 17:36:52 wiz Exp $
d10 1
a10 1
PKGREVISION=	7
@


1.146
log
@*: recursive bump for tiff shlib major bump
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.145 2022/12/28 09:19:53 nia Exp $
d10 1
a10 1
PKGREVISION=	6
@


1.145
log
@asterisk[18,19]: Add missing libedit dependency (included in base on
NetBSD)
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.144 2022/11/23 16:19:32 adam Exp $
d10 1
a10 1
PKGREVISION=	5
@


1.144
log
@massive revision bump after textproc/icu update
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.143 2022/10/26 10:31:17 wiz Exp $
d301 1
@


1.143
log
@*: bump PKGREVISION for libunistring shlib major bump
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.142 2022/06/30 11:18:07 nia Exp $
d10 1
a10 1
PKGREVISION=	4
@


1.142
log
@*: Revbump packages that use Python at runtime without a PKGNAME prefix
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.141 2022/06/28 11:31:06 wiz Exp $
d10 1
a10 1
PKGREVISION=	3
@


1.141
log
@*: recursive bump for perl 5.36
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.140 2022/05/10 11:49:03 nia Exp $
d10 1
a10 1
PKGREVISION=	2
@


1.140
log
@asterisk*: Check for NetBSD properly. Use OPSYS_VERSION.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.139 2022/05/05 08:20:09 nia Exp $
d10 1
a10 1
PKGREVISION=	1
@


1.139
log
@asterisk*: Use OPSYS_VERSION to numerically compare NetBSD versions
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.138 2022/04/18 19:10:33 adam Exp $
d149 4
a152 1
.if exists(/usr/include/unbound.h) && !empty(OS_VERSION:M8.*)
a153 2
.else
PLIST.unbound=	yes
@


1.138
log
@revbump for textproc/icu update
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.137 2021/12/17 08:07:06 jnemeth Exp $
d54 1
a54 2
.if (${OPSYS} == "NetBSD" && \
    (empty(OS_VERSION:M6.99.[4-9]?) && empty(OS_VERSION:M[7-9].*)))
@


1.137
log
@Update to Asterisk 18.9.0.  This is a patch release.

The Asterisk Development Team would like to announce the release
of Asterisk 18.  9.0.

The release of Asterisk 18.9.0 resolves several issues reported by
the community and would have not been possible without your
participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-29720 - res_tonedetect: Add call progress tone detection
      (Reported by N A)
 * ASTERISK-18069 - [patch] app_queue Add Login Time and Last
      Paused Times to Queue Members
      (Reported by Jamuel Starkey)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-29779 - progdocs: Hidden code sections with syntax errors.
      (Reported by Alexander Traud)
 * ASTERISK-29732 - progdocs: Fix grouping for latest Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29771 - Crash occurs when 2 realtime sippeers mysql
      connections are configured and we have a schema warning
      (Reported by Mario Ban)
 * ASTERISK-29776 - stir/shaken: Requires GNU designator
      (Reported by Alexander Traud)
 * ASTERISK-29764 - chan_misdn: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29773 - progdocs: doxyref.h outdated
      (Reported by Alexander Traud)
 * ASTERISK-29765 - xmldoc: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29730 - Segfault in __ao2_ref if refdebug = yes
      (Reported by Alexei Gradinari)
 * ASTERISK-29762 - channels: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29748 - bridging: Infinite loop when both Local
      channel halves in same bridge
      (Reported by Joshua C. Colp)
 * ASTERISK-29754 - odbc: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29753 - parking: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29755 - frame: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29756 - res_ari: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29751 - channel: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29750 - stasis: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29752 - app: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29749 - res_xmpp: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29742 - addons: Fix for Doxygen.
      (Reported by Alexander Traud)
 * ASTERISK-29747 - res_pjsip: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29737 - chan_iax2: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29743 - bridges: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29741 - tests: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29740 - apps: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29733 - progdocs: Avoid name with Doxygen \file
      (Reported by Alexander Traud)
 * ASTERISK-29736 - bridge_channel: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29735 - progdocs: Avoid multiple use of section labels
      (Reported by Alexander Traud)
 * ASTERISK-29734 - progdocs: Use Doxygen \example correctly
      (Reported by Alexander Traud)
 * ASTERISK-29744 - app_morsecode: Fix deadlock
      (Reported by N A)
 * ASTERISK-29703 - res_pjsip_callerid: Fix OLI parsing
      (Reported by N A)
 * ASTERISK-29705 - app_read: Fix custom terminator
      functionality regression
      (Reported by N A)
 * ASTERISK-29724 - BuildSystem: In POSIX sh, == in place of = is undefined.
      (Reported by Alexander Traud)
 * ASTERISK-29702 - sig_analog: Fix truncated buffer copy
      (Reported by N A)
 * ASTERISK-28040 - pbx: "dialplan reload" is removing minus
      symbol from dynamic hints
      (Reported by Daniel Zanutti)
 * ASTERISK-29391 - VoiceMail does not cancel recording on rerecord hangup
      (Reported by N A)
 * ASTERISK-29709 - res_snmp: Not build on recent Debian
      distributions.
      (Reported by Alexander Traud)
 * ASTERISK-29710 - stasis: Clang 13 warns about the unused but
      set variable dispatched.
      (Reported by Alexander Traud)
 * ASTERISK-29711 - aelparse: GCC 11.2 found two maybe uninitialized
      (Reported by Alexander Traud)
 * ASTERISK-29713 - GCC 11.2: two stringop-overread
      (Reported by Alexander Traud)
 * ASTERISK-29682 - Squash compiler issues generated by gcc 11
      (Reported by George Joseph)
 * ASTERISK-29693 - Using --with-crypto and --with-ssl fails on a recompile
      (Reported by George Joseph)
 * ASTERISK-27816 - func_talkdetect's logic is completely broken
      (Reported by Moritz Fain)
 * ASTERISK-29691 - stun: Not all users provide a dst to ast_stun_request
      (Reported by Dennis Haney)
 * ASTERISK-26497 - make install downloads x86_32 variants of
      external modules on non Intel architectures
      (Reported by Corey Farrell)

Improvements made in this release:
-----------------------------------
 * ASTERISK-29777 - documentation: Standardize example syntax
      (Reported by N A)
 * ASTERISK-29715 - app_voicemail: Refactor email generation functions
      (Reported by N A)
 * ASTERISK-29727 - Add type for JSON stasis message RTCP Report
      Received/Sent
      (Reported by Boris P. Korzun)
 * ASTERISK-29714 - Spelling errors
      (Reported by Josh Soref)
 * ASTERISK-29707 - chan_iax2: Allow both key and secret to be
      specified at dial time
      (Reported by N A)
 * ASTERISK-29662 - Add mix option to Playback application for
      say and filename
      (Reported by Shloime Rosenblum)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.136 2021/12/08 16:03:34 adam Exp $
d10 1
@


1.136
log
@revbump for icu and libffi
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.135 2021/11/09 12:04:43 nia Exp $
d8 1
a8 1
DISTNAME=	asterisk-18.8.0
a9 1
PKGREVISION=	1
@


1.135
log
@mk: For consistency, rename PKG_HAS_ to OPSYS_HAVE_.

Requested by jperkin.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.134 2021/11/09 11:11:08 nia Exp $
d10 1
@


1.134
log
@asterisk*: Detect kqueue/timerfd through pkgsrc infrastructure.

Fixes PLIST on NetBSD/current.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.133 2021/11/08 00:58:50 jnemeth Exp $
d122 1
a122 1
.if defined(PKG_HAVE_KQUEUE)
d143 1
a143 1
.if defined(PKG_HAVE_TIMERFD)
@


1.133
log
@Update to Asterisk 18.8.0:

The Asterisk Development Team would like to announce the release
of Asterisk 18.8.0.

The release of Asterisk 18.8.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-29656 - Add CHANNEL_EXISTS function
      (Reported by N A)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-20219 - [patch] - IAX2 Call Encryption Fails with
      RSA authentication
      (Reported by Michael Munger)
 * ASTERISK-29402 - res_pjsip_t38: Socket is bound to IPv4/IPv6
      but platform does not support it
      (Reported by Matthew Kern)
 * ASTERISK-29673 - app_read: Fix null pointer crash regression
      (Reported by N A)
 * ASTERISK-29671 - res_rtp_asterisk: memory leak
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-29668 - ari: Listing bridges fails when dialing
      bridge exists
      (Reported by Joshua C. Colp)
 * ASTERISK-29663 - messaging: AMI MessageSend does not support
      same parameters as dialplan application
      (Reported by Brian J. Murrell)
 * ASTERISK-29578 - app_queue: Custom device state using
      included hints do not update
      (Reported by N A)
 * ASTERISK-29660 - Build failure when disabling PJSIP support
      (Reported by Guido Falsi)

Improvements made in this release:
-----------------------------------
 * ASTERISK-29637 - Add support for future dates in Say.c
      (Reported by Shloime Rosenblum)
 * ASTERISK-29525 - PJSIP remove_existing unavailable contacts
      (Reported by Joseph Nadiv)
 * ASTERISK-29661 - func_vmcount: Add support for multiple
      mailboxes
      (Reported by N A)
 * ASTERISK-29275 - Support of MIME-type for wav16
      (Reported by Boris P. Korzun)
 * ASTERISK-29529 - Add custom logging level
      (Reported by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.132 2021/10/12 03:04:10 jnemeth Exp $
d122 1
a122 1
.if exists(/usr/include/sys/event.h)
d143 1
a143 1
.if ${OPSYS} == "Linux"
@


1.132
log
@Update to Asterisk 18.7.1:

The Asterisk Development Team would like to announce the release
of Asterisk 18.7.1.

The release of Asterisk 18.7.1 resolves an issue reported by the
community and would have not been possible without your participation.

Thank you!

The following issue is resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-29685 - pbx_ael: Infinite loop on reload

      (Reported by Joshua C. Colp)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.131 2021/10/09 07:52:15 jnemeth Exp $
d8 1
a8 1
DISTNAME=	asterisk-18.7.1
@


1.131
log
@Update asterisk to 18.7.0:

The Asterisk Development Team would like to announce the release
of Asterisk 18.7.0.

The release of Asterisk 18.7.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Deprecations made in this release:
-----------------------------------
 * ASTERISK-29548 - app_meetme: Deprecated in 19, to be removed in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29549 - app_osploop: Deprecated in 19, to be removed in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29550 - chan_alsa: Deprecated in 19, to be removed in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29551 - chan_mgcp: Deprecated in 19, to be removed in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29552 - chan_skinny: Deprecated in 19, to be removed in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29553 - res_pktccops: Deprecated in 19, to be removed in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29554 - cdr_mysql: Deprecated in 1.8, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29555 - app_mysql: Deprecated in 1.8, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29557 - app_ices: Deprecated in 16, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29558 - app_macro: Deprecated in 16, to be removed in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29559 - app_fax: Deprecated in 16, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29560 - app_url: Deprecated in 16, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29561 - app_image: Deprecated in 16, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29562 - app_nbscat: Deprecated in 16, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29563 - app_dahdiras: Deprecated in 16, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29564 - cdr_syslog: Deprecated in 16, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29565 - chan_oss: Deprecated in 16, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29566 - chan_phone: Deprecated in 16, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29567 - chan_sip: Deprecated in 17, to be removed in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29568 - chan_nbs: Deprecated in 16, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29569 - chan_misdn: Deprecated in 16, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29570 - chan_vpb: Deprecated in 16, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29571 - res_config_sqlite: Deprecated in 16, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29572 - res_monitor: Deprecated in 16, to be removed in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29573 - conf2ael: Deprecated in 16, to be removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29574 - muted: Deprecated in 16, to be removed in 19
      (Reported by Joshua C. Colp)

New Features made in this release:
-----------------------------------
 * ASTERISK-29496 - Add SendMF application
      (Reported by N A)
 * ASTERISK-29627 - Add STRBETWEEN function
      (Reported by N A)
 * ASTERISK-29628 - Add file and directory functions
      (Reported by N A)
 * ASTERISK-29531 - Add SAYFILES function
      (Reported by N A)
 * ASTERISK-29546 - Add tone detection module
      (Reported by N A)
 * ASTERISK-18454 - Option for Read to be able to accept #
      (Reported by Sta Retji)
 * ASTERISK-29542 - Add audio scrambler
      (Reported by N A)
 * ASTERISK-29478 - Function to drop frames in the TX or RX
      directions
      (Reported by N A)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-29673 - app_read: Fix null pointer crash regression
      (Reported by N A)
 * ASTERISK-29660 - Build failure when disabling PJSIP support
      (Reported by Guido Falsi)
 * ASTERISK-29635 - MP3Player don't work with actual mpg123 versions
      (Reported by Carlos Oliva)
 * ASTERISK-29654 - pjproject includes trailing whitespace in
      sdp format attributes
      (Reported by George Joseph)
 * ASTERISK-29629 - ARI external media channel creation doesn't
      set option data
      (Reported by sungtae kim)
 * ASTERISK-27176 - test_abstract_jb: frames leak
      (Reported by Corey Farrell)
 * ASTERISK-29634 - res_snmp: gcc 11 needs -fPIC to compile correctly
      (Reported by George Joseph)
 * ASTERISK-29630 - Asterisk is unable to read extended number
      format terminfo files
      (Reported by Sean Bright)
 * ASTERISK-28004 - dns: Core ast_dns_get_nameservers does not
      support configured IPv6 servers
      (Reported by Isaac McDonald)
 * ASTERISK-29618 - ConfBridge errors on creation conference room
      (Reported by Alexander Zharov)
 * ASTERISK-29622 - ARI: external media create doesn't use body parameter
      (Reported by sungtae kim)
 * ASTERISK-29614 - app_agent_pool: XML Doc: unterminated entity reference
      (Reported by Alexander Traud)
 * ASTERISK-29609 - Subsequent 'ael reload' will cause a lock up
      (Reported by Mark Murawski)
 * ASTERISK-28701 - app_queue: Core reload resets queue stats,
      even when keepstats=yes
      (Reported by Luke Escude)
 * ASTERISK-29616 - res_rtp_asterisk: sqrt(.) requires the header math.h.
      (Reported by Alexander Traud)
 * ASTERISK-29518 - sig_analog: FCG_CAMA fails to signal ANI
      spill when using MF signaling
      (Reported by Sarah Autumn)
 * ASTERISK-29582 - res_pjproject: Can't map pjproject log
      messages to Asterisk TRACE
      (Reported by George Joseph)
 * ASTERISK-29575 - app_milliwatt: Milliwatt application doesn't
      use the proper timings
      (Reported by N A)
 * ASTERISK-20339 - chan_mgcp, res_pktccops ast_debug support
      (Reported by Tomas Maldonado)
 * ASTERISK-29540 - aelparse: include of context with timings fails
      (Reported by Alexander Traud)
 * ASTERISK-29539 - Segmentation fault at ast_writestream() when
      write handler not defined (happens with OGG/Speex)
      (Reported by Ernani Jos?? Camargo Azevedo)

Improvements made in this release:
-----------------------------------
 * ASTERISK-29472 - res_pjsip: OLI/ANI2 support missing
      (Reported by N A)
 * ASTERISK-29626 - app_stack: Include calling location if
      attempting to branch to nonexistent location
      (Reported by N A)
 * ASTERISK-29632 - Add option to Application_VoiceMail to
      suppress instructions only when a custom greeting is present
      (Reported by Charlie Smurthwaite)
 * ASTERISK-29605 - chan_iax2: Add ANI2
      (Reported by N A)
 * ASTERISK-29508 - STUN server address refresh
      (Reported by S??bastien Duthil)
 * ASTERISK-29612 - bridge_basic: Don't throw warning if
      attended transfer is cancelled
      (Reported by N A)
 * ASTERISK-29544 - Media Cache - Delayed remote sound file
      retrieve delays all playbacks
      (Reported by Andre Barbosa)
 * ASTERISK-29495 - Return integer instead of float if response
      is a whole number
      (Reported by N A)
 * ASTERISK-29541 - app_morsecode: Add American Morse code
      (Reported by N A)
 * ASTERISK-29543 - app_originate: Allow specifying codec(s) to use
      (Reported by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
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# $NetBSD: Makefile,v 1.130 2021/09/29 19:00:25 adam Exp $
d8 1
a8 1
DISTNAME=	asterisk-18.7.0
@


1.130
log
@revbump for boost-libs
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.129 2021/09/19 03:59:25 jnemeth Exp $
d8 1
a8 1
DISTNAME=	asterisk-18.6.0
a9 1
PKGREVISION=	1
@


1.129
log
@Update to Asterisk 18.6.0.

The Asterisk Development Team would like to announce the release
of Asterisk 18.6.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 18.6.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Security bugs fixed in this release:
-----------------------------------
 * ASTERISK-29415 - Crash in PJSIP TLS transport
      (Reported by Andrew Yager)
 * ASTERISK-29381 - chan_pjsip: Remote denial of service by an
      authenticated user
      (Reported by Ivan Poddubny)

New Features made in this release:
-----------------------------------
 * ASTERISK-29389 - Add PJSIP_HEADERS() and ability to read
      header by pattern
      (Reported by Igor Goncharovsky)
 * ASTERISK-29477 - Function to asynchronously store digits dialed
      (Reported by N A)
 * ASTERISK-29454 - New application to reload modules
      (Reported by N A)
 * ASTERISK-29444 - Add application to wait for condition
      (Reported by N A)
 * ASTERISK-29442 - app_dial: Expand A option to allow
      announcement playback to caller
      (Reported by N A)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-29494 - cdr_adaptive_odbc: Prevent throwing warnings
      if CDR filtering is used
      (Reported by N A)
 * ASTERISK-29513 - statsd: Remove non-standard metric type Meter
      (Reported by Rijnhard Hessel)
 * ASTERISK-29526 - G729 audio gets corrupted by Asterisk due to smoother
      (Reported by under)
 * ASTERISK-29392 - chan_iax2: Asterisk crashes when queueing
      video with format
      (Reported by Michael Welk)
 * ASTERISK-29507 - STUN timeout is silently delaying calls
      (Reported by S??bastien Duthil)
 * ASTERISK-27871 - Remote URL in playback must end with file extension
      (Reported by Caesar)
 * ASTERISK-29514 - ari: Audiosocket segfault when no data specified
      (Reported by Igor Goncharovsky)
 * ASTERISK-29503 - Updated identify/match syntax not supported
      by config wizard
      (Reported by Sean Bright)
 * ASTERISK-29480 - fixedjitterbuffer contains an un-wrappered
      assert that triggers on a negative time slew
      (Reported by Dan Cropp)
 * ASTERISK-29485 - core: Inband generation of tones for Busy()
      and Congestion() may not occur
      (Reported by Joshua C. Colp)
 * ASTERISK-29479 - [patch] Channels are not put on hold for
      Session Progress with inactive audio
      (Reported by Bernd Zobl)

Improvements made in this release:
-----------------------------------
 * ASTERISK-29528 - Add support for multiple files for agent announcements
      (Reported by N A)
 * ASTERISK-29501 - ARI - Stasis Playback doesn't hangup call
      when processing a list of invalid files
      (Reported by Andre Barbosa)
 * ASTERISK-29464 - ARI - PlaybackFinish skip error events
      (Reported by Andre Barbosa)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.128 2021/08/01 02:41:58 jnemeth Exp $
d10 1
@


1.128
log
@asterisk18:  Update to 18.5.1

The Asterisk Development Team would like to announce security releases for
Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and 16.8-cert10.

These releases are available for immediate download at

https://downloads.asterisk.org/pub/telephony/asterisk/releases

The following security vulnerabilities were resolved in these versions:

* AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver
  When Asterisk receives a re-INVITE without SDP after having sent
  a BYE request a crash will occur. This occurs due to the Asterisk
  channel no longer being present while code assumes it is.

* AST-2021-008: Remote crash when using IAX2 channel driver
  If the IAX2 channel driver receives a packet that contains an

* AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
                handshake
  Depending on the timing, it's possible for Asterisk to crash when
  using a TLS connection if the underlying socket parent/listener
  gets destroyed during the handshake.

For a full list of changes in the current releases, please see the ChangeLogs:

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1

The security advisories are available at:

https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
https://downloads.asterisk.org/pub/security/AST-2021-009.pdf

Thank you for your continued support of Asterisk!
@
text
@d1 1
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# $NetBSD: Makefile,v 1.127 2021/06/27 21:39:27 jnemeth Exp $
d8 1
a8 1
DISTNAME=	asterisk-18.5.1
@


1.127
log
@comms/asterisk18: update to Asterisk 18.5.0.

pkgsrc change: Fix segfault under aarch64 from ryoon for comms/asterisk16.

-----

The Asterisk Development Team would like to announce the release
of Asterisk 18.5.0.

The release of Asterisk 18.5.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-29446 - app_confbridge: New ConfKick application
      (Reported by N A)
 * ASTERISK-29440 - app_confbridge: Allow ConfBridge answer to
      be suppressed
      (Reported by N A)
 * ASTERISK-29431 - Minimum and maximum dialplan functions
      (Reported by N A)
 * ASTERISK-29439 - func_volume: Volume function can't be read
      (Reported by N A)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-29475 - SayNumber triggers WARNING if caller hangs
      up during application execution
      (Reported by N A)
 * ASTERISK-29404 - Consolidate res_pjsip_messaging fixes for
      domain name
      (Reported by George Joseph)
 * ASTERISK-29441 - Core reload making TCP endpoints go offline
      (Reported by Luke Escude)
 * ASTERISK-28237 - "FRACK!, Failed assertion bad magic number"
      happens when unsubscribe an application from an event source
      (Reported by Lucas Tardioli Silveira)
 * ASTERISK-28393 - Multidomain support issue
      (Reported by Andrea Sannucci)
 * ASTERISK-29433 - res_rtp_asterisk: Server reflexive
      candidates use incorrect raddr for RTCP
      (Reported by Chris)
 * ASTERISK-29397 - pjsip: Asterisk isn't tolerant of RFC8760 UASs
      (Reported by George Joseph)
 * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
      in PJSIP NOTIFY event: dialog  XML body
      (Reported by Marco Paland)
 * ASTERISK-29370 - chan_sip does not recognize
      application/hook-flash
      (Reported by N A)
 * ASTERISK-29377 - cpool_release_pool "double free or
      corruption (out)"
      (Reported by Robert Sutton)
 * ASTERISK-29372 - file.c switch does not account for flash
      events
      (Reported by N A)
 * ASTERISK-29358 - chan_pjsip: Trace message for progress is
      output even if frame is not queued
      (Reported by Michael Maier)
 * ASTERISK-29407 - chan_local: Filtering audio formats should
      not occur on removed streams
      (Reported by Joshua C. Colp)
 * ASTERISK-29030 - res_rtp_asterisk: Additional RTP-frame (with
      wrong SSRC) gets inserted when switching from progress to
      established
      (Reported by Matthias Hensler)

Improvements made in this release:
-----------------------------------
 * ASTERISK-29450 - Allow setting channel variables using
      Originate application
      (Reported by N A)
 * ASTERISK-29459 - Missing configuration from PJSIP to SIP
      conversion script
      (Reported by N A)
 * ASTERISK-29460 - Recognize application/hook-flash in PJSIP
      (Reported by N A)
 * ASTERISK-29434 - Asterisk reveals pjproject version in STUN packets
      (Reported by Jeremy Lain??)
 * ASTERISK-29349 - Silent voicemail option is not completely silent
      (Reported by N A)
 * ASTERISK-29380 - Add Flash AMI event to handle flash events
      (Reported by N A)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0

Thank you for your continued support of Asterisk!
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@
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@d1 1
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# $NetBSD: Makefile,v 1.126 2021/06/13 07:57:52 jnemeth Exp $
d8 1
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DISTNAME=	asterisk-18.5.0
@


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log
@resolve merge conflicts
@
text
@d1 1
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# $NetBSD: Makefile,v 1.30 2021/01/17 08:32:40 jnemeth Exp $
d8 1
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DISTNAME=	asterisk-18.4.0
@


1.125
log
@Deleting comms/asterisk18 (Asterisk 1.8.*) as mentioned on
pkgsrc-users@@ a few weeks ago.  This package is ancient and has
been EOL for a couple of years.  It likely has numerous security
issues.  Also, the PKGNAME will conflict with the upcoming Asterisk
18.* in a couple of years times.  There were no objections.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.124 2018/04/29 21:31:29 adam Exp $
d4 3
a6 1
#       tarballs need to be checked
d8 2
a9 5
DISTNAME=	asterisk-1.8.32.3
DIST_SUBDIR=	${PKGNAME_NOREV}
DISTFILES=	${DEFAULT_DISTFILES}
EXTRACT_ONLY=	${DISTNAME}.tar.gz
PKGREVISION=	25
d14 2
a16 2
OWNER=		jnemeth@@NetBSD.org
HOMEPAGE=	http://www.asterisk.org/
d20 9
d33 1
a33 1
USE_TOOLS+=		bison gmake perl:run pkg-config tar
d35 4
d42 3
d52 9
a60 1
CONFIGURE_ARGS+=	--with-oss=yes
d69 4
a72 4
ASTERISK_USER?=		asterisk
ASTERISK_GROUP?=	asterisk
PKG_GROUPS=		${ASTERISK_GROUP}
PKG_USERS=		${ASTERISK_USER}:${ASTERISK_GROUP}
d74 6
a79 6
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}
d122 1
a122 2
.for dir in ${COMPILER_INCLUDE_DIRS}
.  if exists(${dir}/sys/event.h)
d124 1
a124 2
.  endif
.endfor
d133 1
a133 1
PLIST.mgcp= yes
d138 17
d158 10
a167 1
DISTFILES+=	asterisk-extra-sounds-en-gsm-1.4.15.tar.gz
d172 11
a182 6
SUBST_FILES.configs=	configs/festival.conf.sample
SUBST_FILES.configs+=	configs/http.conf.sample
SUBST_FILES.configs+=	configs/musiconhold.conf.sample
SUBST_FILES.configs+=	configs/osp.conf.sample
SUBST_FILES.configs+=	configs/phoneprov.conf.sample
SUBST_FILES.configs+=	configs/res_config_sqlite.conf.sample
d188 6
a193 5
# XXX gross hack, remove when atomics properly implemented
SUBST_CLASSES.NetBSD+=	atomics
SUBST_STAGE.atomics=	post-configure
SUBST_FILES.atomics=	include/asterisk/autoconfig.h
SUBST_SED.atomics=	-e "s|^\#define HAVE_GCC_ATOMICS 1|\#undef HAVE_GCC_ATOMICS|"
d201 7
d235 1
d237 3
a239 2
.for f in adsi.conf agents.conf ais.conf alarmreceiver.conf alsa.conf	\
	amd.conf app_mysql.conf asterisk.adsi calendar.conf ccss.conf	\
d241 23
a263 18
	cdr_manager.conf cdr_mysql.conf	cdr_odbc.conf cdr_pgsql.conf	\
	cdr_sqlite3_custom.conf cdr_tds.conf cel.conf cel_custom.conf	\
	cel_odbc.conf cel_pgsql.conf cel_sqlite3_custom.conf		\
	cel_tds.conf chan_dahdi.conf chan_mobile.conf chan_ooh323.conf	\
	cli.conf cli_aliases.conf cli_permissions.conf codecs.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 gtalk.conf h323.conf	\
	http.conf iax.conf iaxprov.conf indications.conf jabber.conf	\
	jingle.conf logger.conf manager.conf meetme.conf mgcp.conf	\
	minivm.conf misdn.conf modules.conf musiconhold.conf muted.conf	\
	osp.conf oss.conf phone.conf phoneprov.conf queuerules.conf	\
	queues.conf res_config_mysql.conf res_config_sqlite.conf	\
	res_curl.conf res_fax.conf res_ldap.conf res_odbc.conf		\
	res_pgsql.conf res_pktccops.conf res_snmp.conf			\
	res_stun_monitor.conf rtp.conf say.conf sip.conf udptl.conf	\
	unistim.conf users.conf voicemail.conf vpb.conf
d265 2
a266 1
.endfor
d270 1
a271 3
.if ${OPSYS} == "Darwin"
.  for dir in ${COMPILER_INCLUDE_DIRS}
.    if exists(${dir}/sys/poll.h)
a272 2
.    endif
.  endfor
d277 1
a277 1
	${TAR} xzf ${WRKSRC}/sounds/asterisk-core-sounds-en-gsm-1.4.26.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
d279 1
a279 1
	${TAR} xzf ${DISTDIR}/${DIST_SUBDIR}/asterisk-extra-sounds-en-gsm-1.4.15.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
d286 2
a287 5
	${INSTALL_DATA} ${WRKSRC}/README ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/README-SERIOUSLY.bestpractices.txt ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/UPGRADE-1.2.txt ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/UPGRADE-1.4.txt ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/UPGRADE-1.6.txt ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
d293 5
a297 2
	${INSTALL_DATA} ${WRKSRC}/doc/api-1.6.2-changes.txt ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	cp -r ${WRKSRC}/doc/Asterisk-Admin-Guide ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}/Asterisk-Admin-Guide
d300 1
d302 2
a306 1
.include "../../mk/oss.buildlink3.mk"
d309 1
@


1.124
log
@revbump for boost-libs update
@
text
@d1 1
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# $NetBSD: Makefile,v 1.123 2018/04/16 14:34:15 wiz Exp $
@


1.123
log
@Recursive bump for new fribidi dependency in pango.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.122 2018/04/14 07:34:12 adam Exp $
d10 1
a10 1
PKGREVISION=	24
@


1.122
log
@revbump after icu update
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.121 2018/03/12 11:16:12 wiz Exp $
d10 1
a10 1
PKGREVISION=	23
@


1.121
log
@Recursive bumps for fontconfig and libzip dependency changes.
@
text
@d1 1
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# $NetBSD: Makefile,v 1.120 2018/01/01 21:18:17 adam Exp $
d10 1
a10 1
PKGREVISION=	22
@


1.120
log
@Revbump after boost update
@
text
@d1 1
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# $NetBSD: Makefile,v 1.119 2017/11/30 16:45:16 adam Exp $
d10 1
a10 1
PKGREVISION=	21
@


1.119
log
@Revbump after textproc/icu update
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.118 2017/09/18 09:53:12 maya Exp $
d10 1
a10 1
PKGREVISION=	20
@


1.118
log
@revbump for requiring ICU 59.x
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.117 2017/08/24 20:03:07 adam Exp $
d10 1
a10 1
PKGREVISION=	19
@


1.117
log
@Revbump for boost update
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.116 2017/04/30 01:21:30 ryoon Exp $
d10 1
a10 1
PKGREVISION=	18
@


1.116
log
@Recursive revbump from boost update
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.115 2017/04/22 21:03:26 adam Exp $
d10 1
a10 1
PKGREVISION=	17
@


1.115
log
@Revbump after icu update
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.114 2017/02/12 06:25:09 ryoon Exp $
d10 1
a10 1
PKGREVISION=	16
@


1.114
log
@Recursive revbump from fonts/harfbuzz
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.113 2017/02/06 13:55:09 wiz Exp $
d10 1
a10 1
PKGREVISION=	15
@


1.113
log
@Recursive bump for harfbuzz's new graphite2 dependency.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.112 2017/01/19 18:52:04 agc Exp $
d10 1
a10 1
PKGREVISION=	14
@


1.112
log
@Convert all occurrences (353 by my count) of

	MASTER_SITES= 	site1 \
			site2

style continuation lines to be simple repeated

	MASTER_SITES+= site1
	MASTER_SITES+= site2

lines. As previewed on tech-pkg. With thanks to rillig for fixing pkglint
accordingly.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.111 2017/01/01 16:06:06 adam Exp $
d10 1
a10 1
PKGREVISION=	13
@


1.111
log
@Revbump after boost update
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.110 2016/12/04 05:17:19 ryoon Exp $
d12 3
a14 3
MASTER_SITES=	http://downloads.asterisk.org/pub/telephony/asterisk/ \
		http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/ \
		http://downloads.asterisk.org/pub/telephony/sounds/releases/
@


1.110
log
@Recursive revbump from textproc/icu 58.1
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.109 2016/10/07 18:25:40 adam Exp $
d10 1
a10 1
PKGREVISION=	12
@


1.109
log
@Revbump post boost update
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.108 2016/08/03 10:22:35 adam Exp $
d10 1
a10 1
PKGREVISION=	11
@


1.108
log
@Revbump after graphics/gd update
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.107 2016/07/09 06:37:55 wiz Exp $
d10 1
a10 1
PKGREVISION=	10
@


1.107
log
@Bump PKGREVISION for perl-5.24.0 for everything mentioning perl.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.106 2016/04/11 19:01:44 ryoon Exp $
d10 1
a10 1
PKGREVISION=	9
@


1.106
log
@Recursive revbump from textproc/icu 57.1
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.105 2016/03/05 11:28:10 jperkin Exp $
d10 1
a10 1
PKGREVISION=	8
@


1.105
log
@Bump PKGREVISION for security/openssl ABI bump.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.104 2016/02/25 11:32:19 jperkin Exp $
d10 1
a10 1
PKGREVISION=	7
@


1.104
log
@Use OPSYSVARS.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.103 2015/11/02 12:02:23 tnn Exp $
d10 1
a10 1
PKGREVISION=	6
@


1.103
log
@extraneous parenthesis crept in in Darwin conditional
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.102 2015/11/02 00:34:04 tnn Exp $
a135 1
.if (${OPSYS} == "NetBSD")
d137 1
a137 1
SUBST_CLASSES+=		atomics
d143 1
a143 1
SUBST_CLASSES+=		pktinfo
a146 1
.endif
@


1.102
log
@appease pkglint
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.101 2015/11/02 00:04:00 tnn Exp $
d207 1
a207 1
.    if exists(${dir}/sys/poll.h))
@


1.101
log
@Use ${COMPILER_INCLUDE_DIRS} instead of hardcoded /usr/include
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.100 2015/10/10 01:58:01 ryoon Exp $
d99 2
a100 2
.for _dir_ in ${COMPILER_INCLUDE_DIRS}
.  if exists(${_dir_}/sys/event.h)
d108 3
a110 3
.for _dir_ in ${COMPILER_INCLUDE_DIRS}
.  if(exists(${_dir_}/sys/socket.h))
NOSIGPIPE!=	${GREP} SO_NOSIGPIPE ${_dir_}/sys/socket.h || echo ""
d206 2
a207 2
.  for _dir_ in ${COMPILER_INCLUDE_DIRS}
.    if exists(${_dir_}/sys/poll.h))
@


1.100
log
@Recursive revbump from textproc/icu
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.99 2015/08/18 07:31:03 wiz Exp $
d99 2
a100 1
.if exists(/usr/include/sys/event.h)
d102 2
a103 1
.endif
d108 4
a111 2
NOSIGPIPE!=	${GREP} SO_NOSIGPIPE /usr/include/sys/socket.h || echo ""
.if ${NOSIGPIPE} != ""
d113 3
a115 1
.endif
a203 1
.if (${OPSYS} == "Darwin" && exists(/usr/include/sys/poll.h))
d205 3
d209 2
@


1.99
log
@Bump all packages that depend on curses.bui* or terminfo.bui* since they
might incur ncurses dependencies on some platforms, and ncurses just bumped
its shlib.
Some packages were bumped twice now, sorry for that.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.98 2015/06/12 10:48:29 wiz Exp $
d10 1
a10 1
PKGREVISION=	5
@


1.98
log
@Recursive PKGREVISION bump for all packages mentioning 'perl',
having a PKGNAME of p5-*, or depending such a package,
for perl-5.22.0.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.97 2015/04/28 08:48:11 manu Exp $
d10 1
a10 1
PKGREVISION=	4
@


1.97
log
@Fix crash in asterisk18 startup

The added patch fixes startup crash and was submitted upstream.
While there also remove the ban on i386, as it was tested to run fine.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.92 2015/03/15 22:26:26 jnemeth Exp $
d10 1
a10 1
PKGREVISION=	3
@


1.96
log
@Recursive revbump from databases/unixodbc.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.95 2015/04/25 14:20:29 tnn Exp $
d10 1
a10 1
PKGREVISION=	2
a20 3
# known to have issues on i386, block the package until the bug is fixed
BROKEN_ON_PLATFORM=	NetBSD-*-i386

@


1.95
log
@Recursive revbump following MesaLib update, categories a through f.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.94 2015/04/12 03:35:38 jnemeth Exp $
d10 1
a10 1
PKGREVISION=	1
@


1.94
log
@Update to Asterisk 1.8.32.3:  this is a security fix update.

The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11,
12, and 13. The available security releases are released as versions
1.8.28.cert-5, 1.8.32.3, 11.6-cert11, 11.17.1, 12.8.2, 13.1-cert2,
and 13.3.2.

The release of these versions resolves the following security vulnerability:

* AST-2015-003: TLS Certificate Common name NULL byte exploit

  When Asterisk registers to a SIP TLS device and verifies the
  server, Asterisk will accept signed certificates that match a
  common name other than the one Asterisk is expecting if the signed
  certificate has a common name containing a null byte after the
  portion of the common name that Asterisk expected. This potentially
  allows for a man in the middle attack.

For more information about the details of this vulnerability, please read
security advisory AST-2015-003, which was released at the same time as this
announcement.

For a full list of changes in the current releases, please see the Change Logs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3

The security advisory is available at:

* http://downloads.asterisk.org/pub/security/AST-2015-003.pdf

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.93 2015/04/06 08:17:22 adam Exp $
d10 1
@


1.93
log
@Revbump after updating textproc/icu
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.92 2015/03/15 22:26:26 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.32.2
a9 1
PKGREVISION=	1
@


1.92
log
@NOT_FOR_PLATFORM->BROKEN_ON_PLATFORM as requested by dholland@@
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.91 2015/01/29 21:48:07 jnemeth Exp $
d10 1
@


1.92.2.1
log
@Pullup ticket #4736 - requested by manu
comms/asterisk18: security update

Revisions pulled up:
- comms/asterisk18/Makefile                             1.94,1.97 via patch
- comms/asterisk18/distinfo                             1.60-1.61
- comms/asterisk18/patches/patch-main_loader.c          1.1

---
   Module Name:    pkgsrc
   Committed By:   jnemeth
   Date:           Sun Apr 12 03:35:39 UTC 2015

   Modified Files:
           pkgsrc/comms/asterisk18: Makefile distinfo

   Log Message:
   Update to Asterisk 1.8.32.3:  this is a security fix update.

   The Asterisk Development Team has announced security releases for
   Certified Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11,
   12, and 13. The available security releases are released as versions
   1.8.28.cert-5, 1.8.32.3, 11.6-cert11, 11.17.1, 12.8.2, 13.1-cert2,
   and 13.3.2.

   The release of these versions resolves the following security vulnerability:

   * AST-2015-003: TLS Certificate Common name NULL byte exploit

     When Asterisk registers to a SIP TLS device and verifies the
     server, Asterisk will accept signed certificates that match a
     common name other than the one Asterisk is expecting if the signed
     certificate has a common name containing a null byte after the
     portion of the common name that Asterisk expected. This potentially
     allows for a man in the middle attack.

   For more information about the details of this vulnerability, please read
   security advisory AST-2015-003, which was released at the same time as this
   announcement.

   For a full list of changes in the current releases, please see the Change Logs:

   http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3

   The security advisory is available at:

   * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf

   Thank you for your continued support of Asterisk!

---
   Module Name:    pkgsrc
   Committed By:   manu
   Date:           Tue Apr 28 08:48:11 UTC 2015

   Modified Files:
           pkgsrc/comms/asterisk18: Makefile distinfo
   Added Files:
           pkgsrc/comms/asterisk18/patches: patch-main_loader.c

   Log Message:
   Fix crash in asterisk18 startup

   The added patch fixes startup crash and was submitted upstream.
   While there also remove the ban on i386, as it was tested to run fine.
@
text
@d1 1
a1 1
# $NetBSD$
d6 1
a6 1
DISTNAME=	asterisk-1.8.32.3
a9 1
PKGREVISION=	3
d20 3
@


1.91
log
@Update to asterisk 1.8.32.2:  this is a security fix.

pkgsrc change: adapt to splitting up of speex

The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
11.15.1, 12.8.1, and 13.1.1.

The release of these versions resolves the following security vulnerabilities:

* AST-2015-001: File descriptor leak when incompatible codecs are offered

                Asterisk may be configured to only allow specific audio or
                video codecs to be used when communicating with a
                particular endpoint. When an endpoint sends an SDP offer
                that only lists codecs not allowed by Asterisk, the offer
                is rejected. However, in this case, RTP ports that are
                allocated in the process are not reclaimed.

                This issue only affects the PJSIP channel driver in
                Asterisk. Users of the chan_sip channel driver are not
                affected.

* AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability

                CVE-2014-8150 reported an HTTP request injection
                vulnerability in libcURL. Asterisk uses libcURL in its
                func_curl.so module (the CURL() dialplan function), as well
                as its res_config_curl.so (cURL realtime backend) modules.

                Since Asterisk may be configured to allow for user-supplied
                URLs to be passed to libcURL, it is possible that an
                attacker could use Asterisk as an attack vector to inject
                unauthorized HTTP requests if the version of libcURL
                installed on the Asterisk server is affected by
                CVE-2014-8150.

For more information about the details of these vulnerabilities, please read
security advisory AST-2015-001 and AST-2015-002, which were released at the same
time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1

The security advisories are available at:

* http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
* http://downloads.asterisk.org/pub/security/AST-2015-002.pdf

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.90 2014/12/03 01:00:23 jnemeth Exp $
d21 1
a21 1
NOT_FOR_PLATFORM=	NetBSD-*-i386
@


1.90
log
@Update to Asterisk 1.8.32.1: this is a security fix release.

The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
security releases are released as versions 1.8.28-cert3, 11.6-cert8, 1.8.32.1,
11.14.1, 12.7.1, and 13.0.1.

The release of these versions resolves the following security vulnerabilities:

* AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP
  address families

  Many modules in Asterisk that service incoming IP traffic have ACL options
  ("permit" and "deny") that can be used to whitelist or blacklist address
  ranges. A bug has been discovered where the address family of incoming
  packets is only compared to the IP address family of the first entry in the
  list of access control rules. If the source IP address for an incoming
  packet is not of the same address as the first ACL entry, that packet
  bypasses all ACL rules.

* AST-2014-018: Permission Escalation through DB dialplan function

  The DB dialplan function when executed from an external protocol, such as AMI,
  could result in a privilege escalation. Users with a lower class authorization
  in AMI can access the internal Asterisk database without the required SYSTEM
  class authorization.

For more information about the details of these vulnerabilities, please read
security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015,
AST-2014-016, AST-2014-017, and AST-2014-018, which were released at the same
time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf
 * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.89 2014/11/19 08:30:56 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.32.1
@


1.89
log
@Update to Asterisk 1.8.32.0: this is mostly a bug fix release.

The Asterisk Development Team has announced the release of Asterisk 1.8.32.0.

The release of Asterisk 1.8.32.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-24348 - Built-in editline tab complete segfault with
      MALLOC_DEBUG (Reported by Walter Doekes)
 * ASTERISK-24335 - [PATCH] Asterisk incorrectly responds 503 to
      INVITE retransmissions of rejected calls (Reported by Torrey
      Searle)
 * ASTERISK-23768 - [patch] Asterisk man page contains a (new)
      unquoted minus sign (Reported by Jeremy LainÃ©)
 * ASTERISK-24357 - [fax] Out of bounds error in update_modem_bits
      (Reported by Jeremy LainÃ©)
 * ASTERISK-22945 - [patch] Memory leaks in chan_sip.c with
      realtime peers (Reported by ibercom)
 * ASTERISK-24390 - astobj2: REF_DEBUG reports false leaks with
      ao2_callback with OBJ_MULTIPLE (Reported by Corey Farrell)
 * ASTERISK-24011 - [patch]safe_asterisk tries to set ulimit -n too
      high on linux systems with lots of RAM (Reported by Michael
      Myles)
 * ASTERISK-20784 - Failure to receive an ACK to a SIP Re-INVITE
      results in a SIP channel leak (Reported by NITESH BANSAL)
 * ASTERISK-15879 - [patch] Failure to receive an ACK to a SIP
      Re-INVITE results in a SIP channel leak (Reported by Torrey
      Searle)
 * ASTERISK-24406 - Some caller ID strings are parsed differently
      since 11.13.0 (Reported by Etienne Lessard)
 * ASTERISK-24325 - res_calendar_ews: cannot be used with neon 0.30
      (Reported by Tzafrir Cohen)
 * ASTERISK-13797 - [patch] relax badshell tilde test (Reported by
      Tzafrir Cohen)
 * ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE
      (Reported by Paolo Compagnini)
 * ASTERISK-18923 - res_fax_spandsp usage counter is wrong
      (Reported by Grigoriy Puzankin)
 * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout
      (Reported by Dmitry Melekhov)
 * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy
      when sending qualify requests (Reported by Damian Ivereigh)
 * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
      SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
      abelbeck)
 * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
      against libsrtp-1.5.0 (Reported by Patrick Laimbock)
 * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers
      (Reported by Olle Johansson)
 * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by
      Nick Adams)
 * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled
      (Reported by Corey Farrell)
 * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream
      leaks (Reported by Corey Farrell)
 * ASTERISK-24307 - Unintentional memory retention in stringfields
      (Reported by Etienne Lessard)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.32.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.88 2014/10/14 03:35:05 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.32.0
@


1.88
log
@Update Asterisk to 1.8.31.0.  This is mostly a bugfix release:

The Asterisk Development Team has announced the release of Asterisk 1.8.31.0.

The release of Asterisk 1.8.31.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-24032 - Gentoo compilation emits warning:
      "_FORTIFY_SOURCE" redefined (Reported by Kilburn)
 * ASTERISK-24225 - Dial option z is broken (Reported by
      dimitripietro)
 * ASTERISK-24178 - [patch]fromdomainport used even if not set
      (Reported by Elazar Broad)
 * ASTERISK-24019 - When a Music On Hold stream starts it restarts
      at beginning of file. (Reported by Jason Richards)
 * ASTERISK-24211 - testsuite: Fix the dial_LS_options test
      (Reported by Matt Jordan)
 * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash
      Mohod)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24171 - [patch] Provide a manpage for the aelparse
      utility (Reported by Jeremy LainÃ©)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.31.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.87 2014/10/07 16:47:19 adam Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.31.0
@


1.87
log
@Revbump after updating libwebp and icu
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.86 2014/08/28 01:26:25 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.30.0
a9 1
PKGREVISION=	1
d116 1
a116 1
DISTFILES+=	asterisk-extra-sounds-en-gsm-1.4.14.tar.gz
d207 1
a207 1
	${TAR} xzf ${WRKSRC}/sounds/asterisk-core-sounds-en-gsm-1.4.25.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
d209 1
a209 1
	${TAR} xzf ${DISTDIR}/${DIST_SUBDIR}/asterisk-extra-sounds-en-gsm-1.4.14.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
@


1.86
log
@Update to Asterisk 1.8.30.0:  this is mainly a bugfix update.

The Asterisk Development Team has announced the release of Asterisk 1.8.30.0.

The release of Asterisk 1.8.30.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-23911 - URIENCODE/URIDECODE: WARNING about passing an
      empty string is a bit over zealous (Reported by Matt Jordan)
 * ASTERISK-23814 - No call started after peer dialed (Reported by
      Igor Goncharovsky)
 * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy
      should not call sip_destroy (Reported by Corey Farrell)
 * ASTERISK-23818 - PBX_Lua: after asterisk startup module is
      loaded, but dialplan not available (Reported by Dennis Guse)
 * ASTERISK-18345 - [patch] sips connection dropped by asterisk
      with a large INVITE (Reported by Stephane Chazelas)
 * ASTERISK-23508 - Memory Corruption in
      __ast_string_field_ptr_build_va (Reported by Arnd Schmitter)

Improvements made in this release:
-----------------------------------
 * ASTERISK-21178 - Improve documentation for manager command
      Getvar, Setvar (Reported by Rusty Newton)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.30.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.85 2014/07/26 07:03:26 jnemeth Exp $
d10 1
@


1.86.2.1
log
@Pullup ticket #4566 - requested by jnemeth
comms/asterisk:: security update
comms/asterisk18: security update

Revisions pulled up:
- comms/asterisk/Makefile                                       1.113-1.115
- comms/asterisk/PLIST                                          1.9
- comms/asterisk/distinfo                                       1.67-1.69
- comms/asterisk/patches/patch-contrib_scripts_autosupport      deleted
- comms/asterisk18/Makefile                                     1.88-1.90
- comms/asterisk18/PLIST                                        1.25
- comms/asterisk18/distinfo                                     1.56-1.58

---
   Module Name:	pkgsrc
   Committed By:	jnemeth
   Date:		Tue Oct 14 03:35:05 UTC 2014

   Modified Files:
   	pkgsrc/comms/asterisk18: Makefile PLIST distinfo

   Log Message:
   Update Asterisk to 1.8.31.0.  This is mostly a bugfix release:

   The Asterisk Development Team has announced the release of Asterisk 1.8.31.0.

   The release of Asterisk 1.8.31.0 resolves several issues reported by the
   community and would have not been possible without your participation.
   Thank you!

   The following are the issues resolved in this release:

   Bugs fixed in this release:
   -----------------------------------
    * ASTERISK-24032 - Gentoo compilation emits warning:
         "_FORTIFY_SOURCE" redefined (Reported by Kilburn)
    * ASTERISK-24225 - Dial option z is broken (Reported by
         dimitripietro)
    * ASTERISK-24178 - [patch]fromdomainport used even if not set
         (Reported by Elazar Broad)
    * ASTERISK-24019 - When a Music On Hold stream starts it restarts
         at beginning of file. (Reported by Jason Richards)
    * ASTERISK-24211 - testsuite: Fix the dial_LS_options test
         (Reported by Matt Jordan)
    * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash
         Mohod)

   Improvements made in this release:
   -----------------------------------
    * ASTERISK-24171 - [patch] Provide a manpage for the aelparse
         utility (Reported by Jeremy LainÃ©)

   For a full list of changes in this release, please see the ChangeLog:

   http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.31.0

   Thank you for your continued support of Asterisk!

---
   Module Name:	pkgsrc
   Committed By:	jnemeth
   Date:		Tue Oct 14 03:36:40 UTC 2014

   Modified Files:
   	pkgsrc/comms/asterisk: Makefile PLIST distinfo

   Log Message:
   Update Asterisk to 11.13.0.  This is mostly a bugfix release:

   The Asterisk Development Team has announced the release of Asterisk 11.13.0.

   The release of Asterisk 11.13.0 resolves several issues reported by the
   community and would have not been possible without your participation.
   Thank you!

   The following are the issues resolved in this release:

   Bugs fixed in this release:
   -----------------------------------
    * ASTERISK-24032 - Gentoo compilation emits warning:
         "_FORTIFY_SOURCE" redefined (Reported by Kilburn)
    * ASTERISK-24225 - Dial option z is broken (Reported by
         dimitripietro)
    * ASTERISK-24178 - [patch]fromdomainport used even if not set
         (Reported by Elazar Broad)
    * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload
         warnings and ref leaks (Reported by Walter Doekes)
    * ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP
         ICE candidates in SDP answer (Reported by Badalian Vyacheslav)
    * ASTERISK-24019 - When a Music On Hold stream starts it restarts
         at beginning of file. (Reported by Jason Richards)
    * ASTERISK-23767 - [patch] Dynamic IAX2 registration stops trying
         if ever not able to resolve (Reported by David Herselman)
    * ASTERISK-24211 - testsuite: Fix the dial_LS_options test
         (Reported by Matt Jordan)
    * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash
         Mohod)
    * ASTERISK-23577 - res_rtp_asterisk: Crash in
         ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by
         Jay Jideliov)
    * ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10)
         concurrent WebRTC (avpg/encryption/icesupport) calls (Reported
         by Roman Skvirsky)
    * ASTERISK-24301 - Security: Out of call MESSAGE requests
         processed via Message channel driver can crash Asterisk
         (Reported by Matt Jordan)

   Improvements made in this release:
   -----------------------------------
    * ASTERISK-24171 - [patch] Provide a manpage for the aelparse
         utility (Reported by Jeremy LainÃ©)

   For a full list of changes in this release, please see the ChangeLog:

   http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.13.0

   Thank you for your continued support of Asterisk!

---
   Module Name:	pkgsrc
   Committed By:	jnemeth
   Date:		Wed Nov 19 08:30:57 UTC 2014

   Modified Files:
   	pkgsrc/comms/asterisk18: Makefile distinfo

   Log Message:
   Update to Asterisk 1.8.32.0: this is mostly a bug fix release.

   The Asterisk Development Team has announced the release of Asterisk 1.8.32.0.

   The release of Asterisk 1.8.32.0 resolves several issues reported by the
   community and would have not been possible without your participation.
   Thank you!

   The following are the issues resolved in this release:

   Bugs fixed in this release:
   -----------------------------------
    * ASTERISK-24348 - Built-in editline tab complete segfault with
         MALLOC_DEBUG (Reported by Walter Doekes)
    * ASTERISK-24335 - [PATCH] Asterisk incorrectly responds 503 to
         INVITE retransmissions of rejected calls (Reported by Torrey
         Searle)
    * ASTERISK-23768 - [patch] Asterisk man page contains a (new)
         unquoted minus sign (Reported by Jeremy LainÃ©)
    * ASTERISK-24357 - [fax] Out of bounds error in update_modem_bits
         (Reported by Jeremy LainÃ©)
    * ASTERISK-22945 - [patch] Memory leaks in chan_sip.c with
         realtime peers (Reported by ibercom)
    * ASTERISK-24390 - astobj2: REF_DEBUG reports false leaks with
         ao2_callback with OBJ_MULTIPLE (Reported by Corey Farrell)
    * ASTERISK-24011 - [patch]safe_asterisk tries to set ulimit -n too
         high on linux systems with lots of RAM (Reported by Michael
         Myles)
    * ASTERISK-20784 - Failure to receive an ACK to a SIP Re-INVITE
         results in a SIP channel leak (Reported by NITESH BANSAL)
    * ASTERISK-15879 - [patch] Failure to receive an ACK to a SIP
         Re-INVITE results in a SIP channel leak (Reported by Torrey
         Searle)
    * ASTERISK-24406 - Some caller ID strings are parsed differently
         since 11.13.0 (Reported by Etienne Lessard)
    * ASTERISK-24325 - res_calendar_ews: cannot be used with neon 0.30
         (Reported by Tzafrir Cohen)
    * ASTERISK-13797 - [patch] relax badshell tilde test (Reported by
         Tzafrir Cohen)
    * ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE
         (Reported by Paolo Compagnini)
    * ASTERISK-18923 - res_fax_spandsp usage counter is wrong
         (Reported by Grigoriy Puzankin)
    * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout
         (Reported by Dmitry Melekhov)
    * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy
         when sending qualify requests (Reported by Damian Ivereigh)
    * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
         SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
         abelbeck)
    * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
         against libsrtp-1.5.0 (Reported by Patrick Laimbock)
    * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers
         (Reported by Olle Johansson)
    * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by
         Nick Adams)
    * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled
         (Reported by Corey Farrell)
    * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream
         leaks (Reported by Corey Farrell)
    * ASTERISK-24307 - Unintentional memory retention in stringfields
         (Reported by Etienne Lessard)

   For a full list of changes in this release, please see the ChangeLog:

   http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.32.0

   Thank you for your continued support of Asterisk!

---
   Module Name:	pkgsrc
   Committed By:	jnemeth
   Date:		Wed Nov 19 08:32:48 UTC 2014

   Modified Files:
   	pkgsrc/comms/asterisk: Makefile distinfo
   Removed Files:
   	pkgsrc/comms/asterisk/patches: patch-contrib_scripts_autosupport

   Log Message:
   Update to Asterisk 11.14.0: this is mostly a bugfix release.

   The Asterisk Development Team has announced the release of Asterisk 11.14.0.

   The release of Asterisk 11.14.0 resolves several issues reported by the
   community and would have not been possible without your participation.
   Thank you!

   The following are the issues resolved in this release:

   Bugs fixed in this release:
   -----------------------------------
    * ASTERISK-24348 - Built-in editline tab complete segfault with
         MALLOC_DEBUG (Reported by Walter Doekes)
    * ASTERISK-24335 - [PATCH] Asterisk incorrectly responds 503 to
         INVITE retransmissions of rejected calls (Reported by Torrey
         Searle)
    * ASTERISK-23768 - [patch] Asterisk man page contains a (new)
         unquoted minus sign (Reported by Jeremy LainÃ©)
    * ASTERISK-24357 - [fax] Out of bounds error in update_modem_bits
         (Reported by Jeremy LainÃ©)
    * ASTERISK-20567 - bashism in autosupport (Reported by Tzafrir
         Cohen)
    * ASTERISK-22945 - [patch] Memory leaks in chan_sip.c with
         realtime peers (Reported by ibercom)
    * ASTERISK-24384 - chan_motif: format capabilities leak on module
         load error (Reported by Corey Farrell)
    * ASTERISK-24385 - chan_sip: process_sdp leaks on an error path
         (Reported by Corey Farrell)
    * ASTERISK-24378 - Release AMI connections on shutdown (Reported
         by Corey Farrell)
    * ASTERISK-24354 - AMI sendMessage closes AMI connection on error
         (Reported by Peter Katzmann)
    * ASTERISK-24390 - astobj2: REF_DEBUG reports false leaks with
         ao2_callback with OBJ_MULTIPLE (Reported by Corey Farrell)
    * ASTERISK-24326 - res_rtp_asterisk: ICE-TCP candidates are
         incorrectly attempted (Reported by Joshua Colp)
    * ASTERISK-24011 - [patch]safe_asterisk tries to set ulimit -n too
         high on linux systems with lots of RAM (Reported by Michael
         Myles)
    * ASTERISK-24383 - res_rtp_asterisk: Crash if no candidates
         received for component (Reported by Kevin Harwell)
    * ASTERISK-20784 - Failure to receive an ACK to a SIP Re-INVITE
         results in a SIP channel leak (Reported by NITESH BANSAL)
    * ASTERISK-15879 - [patch] Failure to receive an ACK to a SIP
         Re-INVITE results in a SIP channel leak (Reported by Torrey
         Searle)
    * ASTERISK-24406 - Some caller ID strings are parsed differently
         since 11.13.0 (Reported by Etienne Lessard)
    * ASTERISK-24325 - res_calendar_ews: cannot be used with neon 0.30
         (Reported by Tzafrir Cohen)
    * ASTERISK-13797 - [patch] relax badshell tilde test (Reported by
         Tzafrir Cohen)
    * ASTERISK-22791 - asterisk sends Re-INVITE after receiving a BYE
         (Reported by Paolo Compagnini)
    * ASTERISK-18923 - res_fax_spandsp usage counter is wrong
         (Reported by Grigoriy Puzankin)
    * ASTERISK-24392 - res_fax: fax gateway sessions leak (Reported by
         Corey Farrell)
    * ASTERISK-24393 - rtptimeout=0 doesn't disable rtptimeout
         (Reported by Dmitry Melekhov)
    * ASTERISK-23846 - Unistim multilines. Loss of voice after second
         call drops (on a second line). (Reported by Rustam Khankishyiev)
    * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy
         when sending qualify requests (Reported by Damian Ivereigh)
    * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
         SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
         abelbeck)
    * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
         against libsrtp-1.5.0 (Reported by Patrick Laimbock)
    * ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing
         leak (Reported by Corey Farrell)
    * ASTERISK-24430 - missing letter "p" in word response in
         OriginateResponse event documentation (Reported by Dafi Ni)
    * ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by
         Corey Farrell)
    * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers
         (Reported by Olle Johansson)
    * ASTERISK-24304 - asterisk crashing randomly because of unistim
         channel (Reported by dhanapathy sathya)
    * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by
         Nick Adams)
    * ASTERISK-24466 - app_queue: fix a couple leaks to struct
         call_queue (Reported by Corey Farrell)
    * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled
         (Reported by Corey Farrell)
    * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream
         leaks (Reported by Corey Farrell)
    * ASTERISK-24307 - Unintentional memory retention in stringfields
         (Reported by Etienne Lessard)

   For a full list of changes in this release, please see the ChangeLog:

   http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.14.0

   Thank you for your continued support of Asterisk!

---
   Module Name:	pkgsrc
   Committed By:	jnemeth
   Date:		Wed Dec  3 01:00:23 UTC 2014

   Modified Files:
   	pkgsrc/comms/asterisk18: Makefile distinfo

   Log Message:
   Update to Asterisk 1.8.32.1: this is a security fix release.

   The Asterisk Development Team has announced security releases for Certified
   Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
   security releases are released as versions 1.8.28-cert3, 11.6-cert8, 1.8.32.1,
   11.14.1, 12.7.1, and 13.0.1.

   The release of these versions resolves the following security vulnerabilities:

   * AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP
     address families

     Many modules in Asterisk that service incoming IP traffic have ACL options
     ("permit" and "deny") that can be used to whitelist or blacklist address
     ranges. A bug has been discovered where the address family of incoming
     packets is only compared to the IP address family of the first entry in the
     list of access control rules. If the source IP address for an incoming
     packet is not of the same address as the first ACL entry, that packet
     bypasses all ACL rules.

   * AST-2014-018: Permission Escalation through DB dialplan function

     The DB dialplan function when executed from an external protocol, such as AMI,
     could result in a privilege escalation. Users with a lower class authorization
     in AMI can access the internal Asterisk database without the required SYSTEM
     class authorization.

   For more information about the details of these vulnerabilities, please read
   security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015,
   AST-2014-016, AST-2014-017, and AST-2014-018, which were released at the same
   time as this announcement.

   For a full list of changes in the current releases, please see the ChangeLogs:

   http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1

   The security advisories are available at:

    * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf
    * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf

   Thank you for your continued support of Asterisk!

---
   Module Name:	pkgsrc
   Committed By:	jnemeth
   Date:		Wed Dec  3 01:57:37 UTC 2014

   Modified Files:
   	pkgsrc/comms/asterisk: Makefile distinfo

   Log Message:
   Update to Asterisk 11.14.1:  this is a security fix release.

   The Asterisk Development Team has announced security releases for Certified
   Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
   security releases are released as versions 1.8.28-cert3, 11.6-cert8, 1.8.32.1,
   11.14.1, 12.7.1, and 13.0.1.

   The release of these versions resolves the following security vulnerabilities:

   * AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP
     address families

     Many modules in Asterisk that service incoming IP traffic have ACL options
     ("permit" and "deny") that can be used to whitelist or blacklist address
     ranges. A bug has been discovered where the address family of incoming
     packets is only compared to the IP address family of the first entry in the
     list of access control rules. If the source IP address for an incoming
     packet is not of the same address as the first ACL entry, that packet
     bypasses all ACL rules.

   * AST-2014-018: Permission Escalation through DB dialplan function

     The DB dialplan function when executed from an external protocol, such as AMI,
     could result in a privilege escalation. Users with a lower class authorization
     in AMI can access the internal Asterisk database without the required SYSTEM
     class authorization.

   In addition, the release of 11.6-cert8 and 11.14.1 resolves the following
   security vulnerability:

   * AST-2014-014: High call load with ConfBridge can result in resource exhaustion

     The ConfBridge application uses an internal bridging API to implement
     conference bridges. This internal API uses a state model for channels within
     the conference bridge and transitions between states as different things
     occur. Unload load it is possible for some state transitions to be delayed
     causing the channel to transition from being hung up to waiting for media. As
     the channel has been hung up remotely no further media will arrive and the
     channel will stay within ConfBridge indefinitely.

   In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves
   the following security vulnerability:

   * AST-2014-017: Permission Escalation via ConfBridge dialplan function and
                   AMI ConfbridgeStartRecord Action

     The CONFBRIDGE dialplan function when executed from an external protocol (such
     as AMI) can result in a privilege escalation as certain options within that
     function can affect the underlying system. Additionally, the AMI
     ConfbridgeStartRecord action has options that would allow modification of the
     underlying system, and does not require SYSTEM class authorization in AMI.

   For a full list of changes in the current releases, please see the ChangeLogs:

   http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1

   The security advisories are available at:

    * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf
    * http://downloads.asterisk.org/pub/security/AST-2014-014.pdf
    * http://downloads.asterisk.org/pub/security/AST-2014-017.pdf
    * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf

   Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD$
d6 1
a6 1
DISTNAME=	asterisk-1.8.32.1
d116 1
a116 1
DISTFILES+=	asterisk-extra-sounds-en-gsm-1.4.15.tar.gz
d207 1
a207 1
	${TAR} xzf ${WRKSRC}/sounds/asterisk-core-sounds-en-gsm-1.4.26.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
d209 1
a209 1
	${TAR} xzf ${DISTDIR}/${DIST_SUBDIR}/asterisk-extra-sounds-en-gsm-1.4.15.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
@


1.85
log
@Update to Asterisk 1.8.29.0: this is a general bug fix release.

The Asterisk Development Team has announced the release of Asterisk 1.8.29.0.

The release of Asterisk 1.8.29.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-22551 - Session timer : UAS (Asterisk) starts counting
      at Invite, UAC starts counting at 200 OK. (Reported by i2045)
 * ASTERISK-23582 - [patch]Inconsistent column length in *odbc
      (Reported by Walter Doekes)
 * ASTERISK-23803 - AMI action UpdateConfig EmptyCat clears all
      categories but the requested one (Reported by zvision)
 * ASTERISK-23035 - ConfBridge with name longer than max (32 chars)
      results in several bridges with same conf_name (Reported by
      IÃ±aki CÃ­vico)
 * ASTERISK-23683 - #includes - wildcard character in a path more
      than one directory deep - results in no config parsing on module
      reload (Reported by tootai)
 * ASTERISK-23827 - autoservice thread doesn't exit at shutdown
      (Reported by Corey Farrell)
 * ASTERISK-23814 - No call started after peer dialed (Reported by
      Igor Goncharovsky)
 * ASTERISK-23673 - Security: DOS by consuming the number of
      allowed HTTP connections. (Reported by Richard Mudgett)
 * ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite
      a DEBUG level of zero (Reported by Rusty Newton)
 * ASTERISK-23766 - [patch] Specify timeout for database write in
      SQLite (Reported by Igor Goncharovsky)
 * ASTERISK-23818 - PBX_Lua: after asterisk startup module is
      loaded, but dialplan not available (Reported by Dennis Guse)
 * ASTERISK-23667 - features.conf.sample is unclear as to which
      options can or cannot be set in the general section (Reported by
      David Brillert)
 * ASTERISK-23790 - [patch] - SIP From headers longer than 256
      characters result in dropped call and 'No closing bracket'
      warnings. (Reported by uniken1)
 * ASTERISK-23908 - [patch]When using FEC error correction,
      asterisk tries considers negative sequence numbers as missing
      (Reported by Torrey Searle)
 * ASTERISK-23921 - refcounter.py uses excessive ram for large refs
      files  (Reported by Corey Farrell)
 * ASTERISK-23948 - REF_DEBUG fails to record ao2_ref against
      objects that were already freed (Reported by Corey Farrell)
 * ASTERISK-23984 - Infinite loop possible in ast_careful_fwrite()
      (Reported by Steve Davies)
 * ASTERISK-23897 - [patch]Change in SETUP ACK handling (checking
      PI) in revision 413765 breaks working environments (Reported by
      Pavel Troller)

Improvements made in this release:
-----------------------------------
 * ASTERISK-23564 - [patch]TLS/SRTP status of channel not currently
      available in a CLI command (Reported by Patrick Laimbock)
 * ASTERISK-23492 - Add option to safe_asterisk to disable
      backgrounding (Reported by Walter Doekes)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.29.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.84 2014/07/02 03:20:43 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.29.0
@


1.84
log
@Update to Asterisk 1.8.28.2: this fixes multiple vulnerabilities and
numerous general bugs.  The vulnerabilities fixed are:  AST-2014-001,
AST-2014-002, and AST-2014-007.

-----

The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The
available security releases are released as versions 1.8.15-cert7,
11.6-cert4, 1.8.28.2, 11.10.2, and 12.3.2.

These releases resolve security vulnerabilities that were previously
fixed in 1.8.15-cert6, 11.6-cert3, 1.8.28.1, 11.10.1, and 12.3.1.
Unfortunately, the fix for AST-2014-007 inadvertently introduced
a regression in Asterisk's TCP and TLS handling that prevented
Asterisk from sending data over these transports. This regression
and the security vulnerabilities have been fixed in the versions
specified in this release announcement.

The security patches for AST-2014-007 have been updated with the
fix for the regression, and are available at
http://downloads.asterisk.org/pub/security

Please note that the release of these versions resolves the following security
vulnerabilities:

* AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP
                Connections

For more information about the details of these vulnerabilities,
please read security advisories AST-2014-005, AST-2014-006,
AST-2014-007, and AST-2014-008, which were released with the previous
versions that addressed these vulnerabilities.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.28.2

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2014-007.pdf

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The
available security releases are released as versions 1.8.15-cert6,
11.6-cert3, 1.8.28.1, 11.10.1, and 12.3.1.

These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of these versions resolves the following issue:

* AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP
                Connections

  Establishing a TCP or TLS connection to the configured HTTP or
  HTTPS port respectively in http.conf and then not sending or
  completing a HTTP request will tie up a HTTP session. By doing
  this repeatedly until the maximum number of open HTTP sessions
  is reached, legitimate requests are blocked.

These issues and their resolutions are described in the security advisories.

For more information about the details of these vulnerabilities,
please read security advisories AST-2014-005, AST-2014-006,
AST-2014-007, and AST-2014-008, which were released at the same
time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.28.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2014-007.pdf

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 1.8.28.0.

The release of Asterisk 1.8.28.0 resolves several issues reported
by the community and would have not been possible without your
participation.  Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-23547 - [patch] app_queue removing callers from queue
      when reloading (Reported by Italo Rossi)
 * ASTERISK-22846 - testsuite: masquerade super test fails on all
      branches (still) (Reported by Matt Jordan)
 * ASTERISK-23546 - CB_ADD_LEN does not do what you'd think
      (Reported by Walter Doekes)
 * ASTERISK-23620 - Code path in app_stack fails to unlock list
      (Reported by Bradley Watkins)
 * ASTERISK-18331 - app_sms failure (Reported by David Woodhouse)
 * ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by
      Krzysztof Chmielewski)
 * ASTERISK-23707 - Realtime Contacts: Apparent mismatch between
      PGSQL database state and Asterisk state (Reported by Mark
      Michelson)
 * ASTERISK-23665 - Wrong mime type for codec H263-1998 (h263+)
      (Reported by Guillaume Maudoux)
 * ASTERISK-22977 - chan_sip+CEL: missing ANSWER and PICKUP event
      for INVITE/w/replaces pickup (Reported by Walter Doekes)
 * ASTERISK-23709 - Regression in Dahdi/Analog/waitfordialtone
      (Reported by Steve Davies)
 * ASTERISK-23650 - Intermittent segfault in string functions
      (Reported by Roel van Meer)

Improvements made in this release:
-----------------------------------
 * ASTERISK-23754 - [patch] Use var/lib directory for log file
      configured in asterisk.conf (Reported by Igor Goncharovsky)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.28.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 1.8.27.0.

The release of Asterisk 1.8.27.0 resolves several issues reported
by the community and would have not been possible without your
participation.  Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-22790 - check_modem_rate() may return incorrect rate
      for V.27 (Reported by Paolo Compagnini)
 * ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in
      sip.conf.sample (Reported by Eugene)
 * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted
      minus signs (Reported by Jeremy LainÃ©)
 * ASTERISK-23046 - Custom CDR fields set during a GoSUB called
      from app_queue are not inserted (Reported by Denis Pantsyrev)
 * ASTERISK-23027 - [patch] Spelling typo "transfered" instead of
      "transferred" (Reported by Jeremy LainÃ©)
 * ASTERISK-23008 - Local channels loose CALLERID name when DAHDI
      channel connects (Reported by Michael Cargile)
 * ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted
      request and request queue may differ - fix for locking (Reported
      by adomjan)
 * ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image
      media offer due to invalid or unsupported syntax (Reported by
      adomjan)
 * ASTERISK-22861 - [patch]Specifying a null time as parameter to
      GotoIfTime or ExecIfTime causes segmentation fault (Reported by
      Sebastian Murray-Roberts)
 * ASTERISK-17837 - extconfig.conf - Maximum Include level (1)
      exceeded (Reported by pz)
 * ASTERISK-22662 - Documentation fix? - queues.conf says
      persistentmembers defaults to yes, it appears to lie (Reported
      by Rusty Newton)
 * ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot
      handle selinux port restrictions (Reported by Corey Farrell)
 * ASTERISK-23220 - STACK_PEEK function with no arguments causes
      crash/core dump (Reported by James Sharp)
 * ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload'
      command multiple times on cli_aliases (Reported by Joel Vandal)
 * ASTERISK-22757 - segfault in res_clialiases.so on reload when
      mapping "module reload" command (Reported by Gareth Blades)
 * ASTERISK-17727 - [patch] TLS doesn't get all certificate chain
      (Reported by LN)
 * ASTERISK-23178 - devicestate.h: device state setting functions
      are documented with the wrong return values (Reported by
      Jonathan Rose)
 * ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if
      res_parking.so is not loaded, or if res_parking.conf has no
      configuration (Reported by CJ Oster)
 * ASTERISK-23069 - Custom CDR variable not recorded when set in
      macro called from app_queue (Reported by Bryan Anderson)
 * ASTERISK-19499 - ConfBridge MOH is not working for transferee
      after attended transfer (Reported by Timo TerÃ¤s)
 * ASTERISK-23261 - [patch]Output mixup in
      ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686)
 * ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR
      variables for subsequent records (Reported by zvision)
 * ASTERISK-23141 - Asterisk crashes on Dial(), in
      pbx_find_extension at pbx.c (Reported by Maxim)
 * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
      to minrate=2400, then res_fax refuse to load (Reported by David
      Brillert)
 * ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
      - probably introduced in 11.7.0 (Reported by OK)
 * ASTERISK-23323 - [patch]chan_sip: missing p->owner checks in
      handle_response_invite (Reported by Walter Doekes)
 * ASTERISK-23382 - [patch]Build System: make -qp can corrupt
      menuselect-tree and related files (Reported by Corey Farrell)
 * ASTERISK-23406 - [patch]Fix typo in "sip show peer" (Reported by
      ibercom)
 * ASTERISK-23310 - bridged channel crashes in bridge_p2p_rtp_write
      (Reported by Jeremy LainÃ©)
 * ASTERISK-23104 - Specifying the SetVar AMI without a Channel
      cause Asterisk to crash (Reported by Joel Vandal)
 * ASTERISK-23383 - Wrong sense test on stat return code causes
      unchanged config check to break with include files. (Reported by
      David Woolley)
 * ASTERISK-17523 - Qualify for static realtime peers does not work
      (Reported by Maciej Krajewski)
 * ASTERISK-21406 - [patch] chan_sip deadlock on monlock between
      unload_module and do_monitor (Reported by Corey Farrell)
 * ASTERISK-23373 - [patch]Security: Open FD exhaustion with
      chan_sip Session-Timers (Reported by Corey Farrell)
 * ASTERISK-23340 - Security Vulnerability: stack allocation of
      cookie headers in loop allows for unauthenticated remote denial
      of service attack (Reported by Matt Jordan)
 * ASTERISK-23488 - Logic error in callerid checksum processing
      (Reported by Russ Meyerriecks)
 * ASTERISK-20841 - fromdomain not honored on outbound INVITE
      request (Reported by Kelly Goedert)
 * ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f)
      at astobj2.c:120 (Reported by Jamuel Starkey)
 * ASTERISK-23509 - [patch]SayNumber for Polish language tries to
      play empty files for numbers divisible by 100 (Reported by
      zvision)
 * ASTERISK-23391 - Audit dialplan function usage of channel
      variable (Reported by Corey Farrell)
 * ASTERISK-23548 - POST to ARI sometimes returns no body on
      success (Reported by Scott Griepentrog)

Improvements made in this release:
-----------------------------------
 * ASTERISK-22980 - [patch]Allow building cdr_radius and cel_radius
      against libfreeradius-client (Reported by Jeremy LainÃ©)
 * ASTERISK-22661 - Unable to exit ChanSpy if spied channel does
      not have a call in progress (Reported by Chris Hillman)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.27.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The
available security releases are released as versions 1.8.15-cert5,
11.6-cert2, 1.8.26.1, 11.8.1, and 12.1.1.

The release of these versions resolve the following issues:

* AST-2014-001: Stack overflow in HTTP processing of Cookie headers.

  Sending a HTTP request that is handled by Asterisk with a large number of
  Cookie headers could overflow the stack.

  Another vulnerability along similar lines is any HTTP request with a
  ridiculous number of headers in the request could exhaust system memory.

* AST-2014-002: chan_sip: Exit early on bad session timers request

  This change allows chan_sip to avoid creation of the channel and
  consumption of associated file descriptors altogether if the inbound
  request is going to be rejected anyway.

These issues and their resolutions are described in the security advisories.

For more information about the details of these vulnerabilities,
please read security advisories AST-2014-001, AST-2014-002,
AST-2014-003, and AST-2014-004, which were released at the same
time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.26.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2014-001.pdf
 * http://downloads.asterisk.org/pub/security/AST-2014-002.pdf

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 1.8.26.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.26.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-22544 - Italian prompt vm-options has advertisement in
      it (Reported by Rusty Newton)
 * ASTERISK-12117 - chan_sip creates a new local tag (from-tag) for
      every register message (Reported by Pawel Pierscionek)
 * ASTERISK-20862 - Asterisk min and max member penalties not
      honored when set with 0 (Reported by Schmooze Com)
 * ASTERISK-22746 - [patch]Crash in chan_dahdi during caller id
      read (Reported by Michael Walton)
 * ASTERISK-22788 - [patch] main/translate.c: access to variable f
      after free in ast_translate() (Reported by Corey Farrell)
 * ASTERISK-21242 - Segfault when T.38 re-invite retransmission
      receives 200 OK (Reported by Ashley Winters)
 * ASTERISK-22590 - BufferOverflow in unpacksms16() when receiving
      16 bit multipart SMS with app_sms (Reported by Jan Juergens)
 * ASTERISK-22905 - Prevent Asterisk functions that are 'dangerous'
      from being executed from external interfaces (Reported by Matt
      Jordan)
 * ASTERISK-23021 - Typos in code : "avaliable" instead of
      "available" (Reported by Jeremy LainÃ©)
 * ASTERISK-22970 - [patch]Documentation fix for QUOTE() (Reported
      by Gareth Palmer)
 * ASTERISK-22856 - [patch]SayUnixTime in polish reads minutes
      instead of seconds (Reported by Robert Mordec)
 * ASTERISK-22854 - [patch] - Deadlock between cel_pgsql unload and
      core_event_dispatcher taskprocessor thread (Reported by Etienne
      Lessard)
 * ASTERISK-22910 - [patch] - REPLACE() calls strcpy on overlapping
      memory when <replace-char> is empty (Reported by Gareth Palmer)
 * ASTERISK-22871 - cel_pgsql module not loading after "reload" or
      "reload cel_pgsql.so" command (Reported by Matteo)
 * ASTERISK-23084 - [patch]rasterisk needlessly prints the
      AST-2013-007 warning (Reported by Tzafrir Cohen)
 * ASTERISK-17138 - [patch] Asterisk not re-registering after it
      receives "Forbidden - wrong password on authentication"
      (Reported by Rudi)
 * ASTERISK-23011 - [patch]configure.ac and pbx_lua don't support
      lua 5.2 (Reported by George Joseph)
 * ASTERISK-22834 - Parking by blind transfer when lot full orphans
      channels (Reported by rsw686)
 * ASTERISK-23047 - Orphaned (stuck) channel occurs during a failed
      SIP transfer to parking space (Reported by Tommy Thompson)
 * ASTERISK-22946 - Local From tag regression with sipgate.de
      (Reported by Stephan Eisvogel)
 * ASTERISK-23010 - No BYE message sent when sip INVITE is received
      (Reported by Ryan Tilton)

Improvements made in this release:
-----------------------------------
 * ASTERISK-22659 - Make a new core and extra sounds release
      (Reported by Rusty Newton)
 * ASTERISK-22918 - dahdi show channels slices PRI channel dnid on
      output (Reported by outtolunc)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.26.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.83 2014/05/29 23:35:20 wiz Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.28.2
@


1.83
log
@Bump for perl-5.20.0.
Do it for all packages that
* mention perl, or
* have a directory name starting with p5-*, or
* depend on a package starting with p5-
like last time, for 5.18, where this didn't lead to complaints.
Let me know if you have any this time.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.82 2014/05/05 00:47:41 ryoon Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.25.0
a9 1
PKGREVISION=	4
d116 1
a116 1
DISTFILES+=	asterisk-extra-sounds-en-gsm-1.4.13.tar.gz
d140 4
a143 4
SUBST_CLASSES+=         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|"
d207 1
a207 1
	${TAR} xzf ${WRKSRC}/sounds/asterisk-core-sounds-en-gsm-1.4.24.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
d209 1
a209 1
	${TAR} xzf ${DISTDIR}/${DIST_SUBDIR}/asterisk-extra-sounds-en-gsm-1.4.13.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
@


1.82
log
@Recursive revbump from x11/pixman
Fix PR pkg/48777
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.81 2014/04/09 07:27:03 obache Exp $
d10 1
a10 1
PKGREVISION=	3
@


1.81
log
@recursive bump from icu shlib major bump.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.80 2014/02/12 23:17:37 tron Exp $
d10 1
a10 1
PKGREVISION=	2
@


1.80
log
@Recursive PKGREVISION bump for OpenSSL API version bump.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.79 2014/01/07 23:33:28 jnemeth Exp $
d10 1
a10 1
PKGREVISION=	1
@


1.79
log
@Update to Asterisk 1.8.25.0: this is a minor bug fix update

The Asterisk Development Team has announced the release of Asterisk 1.8.25.0.

The release of Asterisk 1.8.25.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* --- chan_sip: Fix an issue where an incompatible audio format may be
      added to SDP.

* --- cdr_adaptive_odbc: Also apply a filter when the CDR value is
      empty.

* --- app_queue: Fix Queuelog EXITWITHKEY only logging two of four
      fields

* --- chan_sip: Do not increment the SDP version between 183 and 200
      responses.

* --- chan_sip: Fix Realtime Peer Update Problem When Un-registering
      And Expires Header In 200ok

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.25.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.78 2013/12/17 02:29:11 jnemeth Exp $
d10 1
@


1.78
log
@Update to Asterisk 1.8.24.1:  this is a security update that fixes
AST-2013-006 and AST-2013-007.

The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security
releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4,
10.12.4-digiumphones, and 11.6.1.

The release of these versions resolve the following issues:

* A buffer overflow when receiving odd length 16 bit messages in app_sms. An
  infinite loop could occur which would overwrite memory when a message is
  received into the unpacksms16() function and the length of the message is an
  odd number of bytes.

* Prevent permissions escalation in the Asterisk Manager Interface. Asterisk
  now marks certain individual dialplan functions as 'dangerous', which will
  inhibit their execution from external sources.

  A 'dangerous' function is one which results in a privilege escalation. For
  example, if one were to read the channel variable SHELL(rm -rf /) Bad
  Things(TM) could happen; even if the external source has only read
  permissions.

  Execution from external sources may be enabled by setting 'live_dangerously'
  to 'yes' in the [options] section of asterisk.conf. Although doing so is not
  recommended.

These issues and their resolutions are described in the security advisories.

For more information about the details of these vulnerabilities, please read
security advisories AST-2013-006 and AST-2013-007, which were
released at the same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.24.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf
 * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.77 2013/10/26 22:29:22 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.24.1
@


1.77
log
@Update to Asterisk 1.8.24.0: this is a general bugfix release:

The Asterisk Development Team has announced the release of Asterisk 1.8.24.0.

The release of Asterisk 1.8.24.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* --- Fix a longstanding issue with MFC-R2 configuration that
      prevented users

* --- Fix Not Storing Current Incoming Recv Address

* --- Fix Segfault When Syntax Of A Line Under [applicationmap] Is
      Invalid

* --- Tolerate presence of RFC2965 Cookie2 header by ignoring it

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.24.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.76 2013/10/19 09:07:00 adam Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.24.0
@


1.76
log
@Revbump after updating textproc/icu
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.75 2013/10/10 14:42:02 ryoon Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.23.1
a9 1
PKGREVISION=	3
@


1.75
log
@Recursive revbump from pango-1.36.0
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.74 2013/09/02 19:50:57 adam Exp $
d10 1
a10 1
PKGREVISION=	2
@


1.74
log
@Revbump after cairo update
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.73 2013/08/29 03:00:47 jnemeth Exp $
d10 1
a10 1
PKGREVISION=	1
@


1.73
log
@Update to Asterisk 1.8.23.1: this is a security fix release to fix
AST-2013-004 and AST-2013-005.

The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The
available security rele ases are released as versions 1.8.15-cert2,
11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-di giumphones, and 11.5.1.

The release of these versions resolve the following issues:

* A remotely exploitable crash vulnerability exists in the SIP
  channel driver if an ACK with SDP is received after the channel
  has been terminated.  The handling code incorrectly assumes that
  the channel will always be present.

* A remotely exploitable crash vulnerability exists in the SIP
  channel driver if an invalid SDP is sent in a SIP request that
  defines media descriptions before connection information. The
  handling code incorrectly attempts to reference the socket address
  information even though that information has not yet been set.

These issues and their resolutions are described in the security advisories.

For more information about the details of these vulnerabilities,
please read security advisories AST-2013-004 and AST-2013-005,
which were released at the same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.23.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2013-004.pdf
 * http://downloads.asterisk.org/pub/security/AST-2013-005.pdf

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.72 2013/07/21 17:20:26 jnemeth Exp $
d10 1
@


1.72
log
@Update to Asterisk 1.8.23.0: this is a bugfix release

pkgsrc changes:
- add work around for NetBSD's incompatible implementation of IP_PKTINFO
- core sounds package was updated to 1.4.24

The Asterisk Development Team has announced the release of Asterisk 1.8.23.0.

The release of Asterisk 1.8.23.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* --- Fix a memory copying bug in slinfactory which was causing
      mixmonitor issues.

* --- IAX2: fix race condition with nativebridge transfers.

* --- Fix crash in chan_sip when a core initiated op occurs at the
      same time as a BYE

* --- Fix The Payload Being Set On CN Packets And Do Not Set Marker
      Bit

* --- chan_sip: Session-Expires: Set timer to correctly expire at
      (~2/3) of the interval when not the refresher

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.23.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.71 2013/07/12 10:44:53 jperkin Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.23.0
@


1.71
log
@Bump PKGREVISION of all packages which create users, to pick up change of
sysutils/user_* packages.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.70 2013/06/16 22:10:13 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.22.0
a9 1
PKGREVISION=	6
d132 1
a133 1
.if (${OPSYS} == "NetBSD")
d138 6
d207 1
a207 1
	${TAR} xzf ${WRKSRC}/sounds/asterisk-core-sounds-en-gsm-1.4.23.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
@


1.70
log
@Asterisk is known to fail on 32-bit systems, specifically i386.  Mark it
as such until the bug is found and fixed.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.69 2013/06/15 20:51:10 jnemeth Exp $
d10 1
a10 1
PKGREVISION=	5
@


1.69
log
@- fix compile issue when newlocale support is detected, from joerg@@
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.68 2013/06/06 12:54:09 wiz Exp $
d21 3
@


1.68
log
@Bump PKGREVISION for libXft changes for NetBSD native X support on
NetBSD 6, requested by tron.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.67 2013/06/04 22:15:47 tron Exp $
d10 1
a10 1
PKGREVISION=	4
@


1.67
log
@Try to fix the fallout caused by the fix for PR pkg/47882. Part 3:

Recursively bump package revisions again after the "freetype2" and
"fontconfig" handling was fixed.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.66 2013/06/03 10:04:40 wiz Exp $
d10 1
a10 1
PKGREVISION=	3
@


1.66
log
@Bump freetype2 and fontconfig dependencies to current pkgsrc versions,
to address issues with NetBSD-6(and earlier)'s fontconfig not being
new enough for pango.

While doing that, also bump freetype2 dependency to current pkgsrc
version.

Suggested by tron in PR 47882
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.65 2013/05/31 12:39:42 wiz Exp $
d10 1
a10 1
PKGREVISION=	2
@


1.65
log
@Bump all packages for perl-5.18, that
a) refer 'perl' in their Makefile, or
b) have a directory name of p5-*, or
c) have any dependency on any p5-* package

Like last time, where this caused no complaints.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.64 2013/05/18 01:00:43 jnemeth Exp $
d10 1
a10 1
PKGREVISION=	1
@


1.64
log
@Update to Asterisk 1.8.22.0: this is a bug fix update.

The Asterisk Development Team has announced the release of Asterisk 1.8.22.0.

The release of Asterisk 1.8.22.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* --- Fix Sorting Order For Parking Lots Stored In Static Realtime

* --- Make ParkAndAnnounce return to priority + 1 when return context
      is not defined

* --- When a session timer expires during a T.38 call, re-invite with
      correct SDP

* --- Fix several unreleased mutex locks that cause problem with
      processing calls

* --- Fix crash when AMI redirect action redirects two channels out of
      a bridge.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.22.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.63 2013/05/09 07:39:29 adam Exp $
d10 1
@


1.63
log
@Massive revbump after updating graphics/ilmbase, graphics/openexr, textproc/icu.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.62 2013/04/18 16:55:22 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.21.0
a9 1
PKGREVISION=	1
@


1.62
log
@Update to Asterisk 1.8.21.0: this is a bug fix update

The Asterisk Development Team has announced the release of Asterisk 1.8.21.0.

The release of Asterisk 1.8.21.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* --- Fix issue where chan_mobile fails to bind to first available port

* --- Fix station ringback; trunk hangup issues in SLA

* --- Fix Queue Log Reporting Every Call COMPLETECALLER With "h"
      Extension Present

* --- Fix Record-Route parsing for large headers.

* --- Fix AMI redirect action with two channels failing to redirect
      both channels.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.21.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.61 2013/04/10 05:24:38 jnemeth Exp $
d10 1
@


1.61
log
@Update to Asterisk 1.2.20.2: this is a security update which fixes
AST-2013-001, AST-2013-002, and AST-2013-003.

The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases
are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones,
and 11.2.2.


The release of these versions resolve the following issues:

* A denial of service exists in Asterisk's HTTP server. AST-2012-014, fixed
  in January of this year, contained a fix for Asterisk's HTTP server for a
  remotely-triggered crash. While the fix prevented the crash from being
  triggered, a denial of service vector still exists with that solution if an
  attacker sends one or more HTTP POST requests with very large Content-Length
  values.

  This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11

* A potential username disclosure exists in the SIP channel driver. When
  authenticating a SIP request with alwaysauthreject enabled, allowguest
  disabled, and autocreatepeer disabled, Asterisk discloses whether a user
  exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways.

  This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11

These issues and their resolutions are described in the security advisories.

For more information about the details of these vulnerabilities, please read
security advisories AST-2013-001, AST-2013-002, and AST-2013-003, which were
released at the same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.20.2

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2013-001.pdf
 * http://downloads.asterisk.org/pub/security/AST-2013-002.pdf
 * http://downloads.asterisk.org/pub/security/AST-2013-003.pdf

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.60 2013/02/16 11:20:35 wiz Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.20.2
d113 1
a113 1
DISTFILES+=	asterisk-extra-sounds-en-gsm-1.4.11.tar.gz
d198 1
a198 1
	${TAR} xzf ${WRKSRC}/sounds/asterisk-core-sounds-en-gsm-1.4.22.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
d200 1
a200 1
	${TAR} xzf ${DISTDIR}/${DIST_SUBDIR}/asterisk-extra-sounds-en-gsm-1.4.11.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
@


1.60
log
@Recursive bump for png-1.6.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.59 2013/02/07 02:24:31 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.20.1
a9 1
PKGREVISION=	1
@


1.60.2.1
log
@Pullup ticket #4116 - requested by jnemeth
comms/asterisk: security update
comms/asterisk10: security update
comms/asterisk18: security update

Revisions pulled up:
- comms/asterisk/Makefile                                       1.84
- comms/asterisk/distinfo                                       1.54
- comms/asterisk10/Makefile                                     1.43
- comms/asterisk10/distinfo                                     1.26
- comms/asterisk18/Makefile                                     1.61
- comms/asterisk18/distinfo                                     1.44

---
   Module Name:    pkgsrc
   Committed By:   jnemeth
   Date:           Wed Apr 10 05:24:39 UTC 2013

   Modified Files:
           pkgsrc/comms/asterisk18: Makefile distinfo

   Log Message:
   Update to Asterisk 1.2.20.2: this is a security update which fixes
   AST-2013-001, AST-2013-002, and AST-2013-003.

   The Asterisk Development Team has announced security releases for Certified
   Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases
   are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones,
   and 11.2.2.

   The release of these versions resolve the following issues:

   * A denial of service exists in Asterisk's HTTP server. AST-2012-014, fixed
     in January of this year, contained a fix for Asterisk's HTTP server for a
     remotely-triggered crash. While the fix prevented the crash from being
     triggered, a denial of service vector still exists with that solution if an
     attacker sends one or more HTTP POST requests with very large Content-Length
     values.

     This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11

   * A potential username disclosure exists in the SIP channel driver. When
     authenticating a SIP request with alwaysauthreject enabled, allowguest
     disabled, and autocreatepeer disabled, Asterisk discloses whether a user
     exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways.

     This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11

   These issues and their resolutions are described in the security advisories.

   For more information about the details of these vulnerabilities, please read
   security advisories AST-2013-001, AST-2013-002, and AST-2013-003, which were
   released at the same time as this announcement.

   For a full list of changes in the current releases, please see the ChangeLogs:

   http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.20.2

   The security advisories are available at:

    * http://downloads.asterisk.org/pub/security/AST-2013-001.pdf
    * http://downloads.asterisk.org/pub/security/AST-2013-002.pdf
    * http://downloads.asterisk.org/pub/security/AST-2013-003.pdf

   Thank you for your continued support of Asterisk!

---
   Module Name:    pkgsrc
   Committed By:   jnemeth
   Date:           Wed Apr 10 05:27:08 UTC 2013

   Modified Files:
           pkgsrc/comms/asterisk10: Makefile distinfo

   Log Message:
   Update to Asterisk 10.12.2:  this is a security update which fixes
   AST-2013-001, AST-2013-002, and AST-2013-003.

   The Asterisk Development Team has announced security releases for Certified
   Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases
   are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones,
   and 11.2.2.

   The release of these versions resolve the following issues:

   * A denial of service exists in Asterisk's HTTP server. AST-2012-014, fixed
     in January of this year, contained a fix for Asterisk's HTTP server for a
     remotely-triggered crash. While the fix prevented the crash from being
     triggered, a denial of service vector still exists with that solution if an
     attacker sends one or more HTTP POST requests with very large Content-Length
     values.

     This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11

   * A potential username disclosure exists in the SIP channel driver. When
     authenticating a SIP request with alwaysauthreject enabled, allowguest
     disabled, and autocreatepeer disabled, Asterisk discloses whether a user
     exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways.

     This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11

   These issues and their resolutions are described in the security advisories.

   For more information about the details of these vulnerabilities, please read
   security advisories AST-2013-001, AST-2013-002, and AST-2013-003, which were
   released at the same time as this announcement.

   For a full list of changes in the current releases, please see the ChangeLogs:

   http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.2

   The security advisories are available at:

    * http://downloads.asterisk.org/pub/security/AST-2013-001.pdf
    * http://downloads.asterisk.org/pub/security/AST-2013-002.pdf
    * http://downloads.asterisk.org/pub/security/AST-2013-003.pdf

   Thank you for your continued support of Asterisk!

---
   Module Name:    pkgsrc
   Committed By:   jnemeth
   Date:           Wed Apr 10 05:28:56 UTC 2013

   Modified Files:
           pkgsrc/comms/asterisk: Makefile distinfo

   Log Message:
   Update to Asterisk 11.2.2:  this is a security update which fixes
   AST-2013-001, AST-2013-002, and AST-213-003.

   The Asterisk Development Team has announced security releases for Certified
   Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases
   are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones,
   and 11.2.2.

   The release of these versions resolve the following issues:

   * A possible buffer overflow during H.264 format negotiation. The format
     attribute resource for H.264 video performs an unsafe read against a media
     attribute when parsing the SDP.

     This vulnerability only affected Asterisk 11.

   * A denial of service exists in Asterisk's HTTP server. AST-2012-014, fixed
     in January of this year, contained a fix for Asterisk's HTTP server for a
     remotely-triggered crash. While the fix prevented the crash from being
     triggered, a denial of service vector still exists with that solution if an
     attacker sends one or more HTTP POST requests with very large Content-Length
     values.

     This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11

   * A potential username disclosure exists in the SIP channel driver. When
     authenticating a SIP request with alwaysauthreject enabled, allowguest
     disabled, and autocreatepeer disabled, Asterisk discloses whether a user
     exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways.

     This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11

   These issues and their resolutions are described in the security advisories.

   For more information about the details of these vulnerabilities, please read
   security advisories AST-2013-001, AST-2013-002, and AST-2013-003, which were
   released at the same time as this announcement.

   For a full list of changes in the current releares, please see the ChangeLogs:

   http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.2.2

   The security advisories are available at:

    * http://downloads.asterisk.org/pub/security/AST-2013-001.pdf
    * http://downloads.asterisk.org/pub/security/AST-2013-002.pdf
    * http://downloads.asterisk.org/pub/security/AST-2013-003.pdf

   Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD$
d6 1
a6 1
DISTNAME=	asterisk-1.8.20.2
d10 1
@


1.59
log
@Update to Asterisk 1.8.20.1:  this is a minor bugfix release

----- 1.8.20.1

The Asterisk Development Team has announced the release of Asterisk 1.8.20.1.

The release of Asterisk 1.8.20.1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

* --- Fix astcanary startup problem due to wrong pid value from before
      daemon call

* --- Update init.d scripts to handle stderr; readd splash screen for
      remote consoles

* --- Reset RTP timestamp; sequence number on SSRC change

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.20.1

Thank you for your continued support of Asterisk!

----- 1.8.20.0

The Asterisk Development Team has announced the release of Asterisk 1.8.20.0.

The release of Asterisk 1.8.20.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* --- app_meetme: Fix channels lingering when hung up under certain
      conditions

* --- Fix stuck DTMF when bridge is broken.

* --- Improve Code Readability And Fix Setting natdetected Flag

* --- Fix extension matching with the '-' char.

* --- Fix call files when astspooldir is relative.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.20.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.58 2013/02/06 23:21:50 jperkin Exp $
d10 1
@


1.58
log
@PKGREVISION bumps for the security/openssl 1.0.1d update.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.57 2013/01/26 21:37:11 adam Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.19.1
a9 1
PKGREVISION=	2
@


1.57
log
@Revbump after graphics/jpeg and textproc/icu
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.56 2013/01/03 02:16:17 jnemeth Exp $
d10 1
a10 1
PKGREVISION=	1
@


1.56
log
@de to Asterisk 1.8.19.1;  this is a security fix to fix AST-2012-14
and AST-2012-015.

Approved for commit during freeze by: agc

The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.11 and Asterisk 1.8, 10, and 11. The available security releases
are released as versions 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones,
and 11.1.1.

The release of these versions resolve the following two issues:

* Stack overflows that occur in some portions of Asterisk that manage a TCP
  connection. In SIP, this is exploitable via a remote unauthenticated session;
  in XMPP and HTTP connections, this is exploitable via remote authenticated
  sessions.

* A denial of service vulnerability through exploitation of the device state
  cache. Anonymous calls had the capability to create devices in Asterisk that
  would never be disposed of.

These issues and their resolutions are described in the security advisories.

For more information about the details of these vulnerabilities, please read
security advisories AST-2012-014 and AST-2012-015, which were released at the
same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2012-014.pdf
 * http://downloads.asterisk.org/pub/security/AST-2012-015.pdf

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.55 2012/12/16 01:52:01 obache Exp $
d10 1
@


1.55
log
@recursive bump from cyrus-sasl libsasl2 shlib major bump.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.54 2012/12/14 01:32:00 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.19.0
a9 1
PKGREVISION=	1
@


1.54
log
@Update to Asterisk 1.8.19.0: this is a bugfix release.

----- 1.8.19.0:

The Asterisk Development Team has announced the release of Asterisk 1.8.19.0.

The release of Asterisk 1.8.19.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* --- Prevent resetting of NATted realtime peer address on reload.

* --- Do not use a FILE handle when doing SIP TCP reads.

* --- Fix execution of 'i' extension due to uninitialized variable.

* --- Ensure that the Queue application tracks busy members in off
      nominal situations

* --- Properly extract the Body information of an EWS calendar item

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.19.0

Thank you for your continued support of Asterisk!

----- 1.8.18.1:

The Asterisk Development Team has announced the release of Asterisk 1.8.18.1.

The release of Asterisk 1.8.18.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

* --- chan_local: Fix local_pvt ref leak in local_devicestate().

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.18.1

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.53 2012/11/11 21:29:04 jnemeth Exp $
d10 1
@


1.53
log
@     Update to Asterisk 1.8.18.0:  this is a bug fix release:

The Asterisk Development Team has announced the release of Asterisk 1.8.18.0.

The release of Asterisk 1.8.18.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* --- dsp.c User Configurable DTMF_HITS_TO_BEGIN and
      DTMF_MISSES_TO_END

* --- Fix error where improper IMAP greetings would be deleted.

* --- iax2-provision: Fix improper return on failed cache retrieval

* --- Fix T.38 support when used with chan_local in between.

* --- Fix an issue where media would not flow for situations where the
      legacy STUN code is in use.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.18.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.52 2012/10/08 23:56:47 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.18.0
@


1.52
log
@update to Asterisk 1.8.17.0:  this is a bugfix release:

The Asterisk Development Team has announced the release of Asterisk 1.8.17.0.

The release of Asterisk 1.8.17.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* --- Fix channel reference leak in ChanSpy.

* --- dsp.c: Fix multiple issues when no-interdigit delay is present,
      and fast DTMF 50ms/50ms

* --- Fix bug where final queue member would not be removed from
      memory.

* --- Fix memory leak when CEL is successfully written to PostgreSQL
      database

* --- Fix DUNDi message routing bug when neighboring peer is
      unreachable

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.17.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.51 2012/10/08 23:01:19 adam Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.17.0
@


1.51
log
@Revbump after updating graphics/pango
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.50 2012/10/03 21:54:05 wiz Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.16.0
a9 1
PKGREVISION=	4
@


1.50
log
@Bump all packages that use perl, or depend on a p5-* package, or
are called p5-*.

I hope that's all of them.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.49 2012/10/03 11:24:39 asau Exp $
d10 1
a10 1
PKGREVISION=	3
@


1.49
log
@Drop superfluous PKG_DESTDIR_SUPPORT, "user-destdir" is default these days.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.48 2012/10/02 17:10:42 tron Exp $
d10 1
a10 1
PKGREVISION=	2
@


1.48
log
@Mass recursive bump after the dependence fix of the "cairo" package
requested by Thomas Klausner.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.47 2012/09/15 10:03:57 obache Exp $
a22 2
PKG_DESTDIR_SUPPORT=	user-destdir

@


1.47
log
@recursive bump from libffi shlib major bump
(additionaly, reset PKGREVISION of qt4-* sub packages from base qt4 update)
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.46 2012/09/14 02:26:28 jnemeth Exp $
d10 1
a10 1
PKGREVISION=	1
@


1.47.2.1
log
@Pullup ticket #3927 - requested by tron
graphics/cairo: build fix

Revisions pulled up:
- archivers/ark/Makefile                                        1.9
- archivers/file-roller/Makefile                                1.121
- audio/albumplayer/Makefile                                    1.22
- audio/amarok-kde3/Makefile                                    1.28
- audio/amarok/Makefile                                         1.115
- audio/ario/Makefile                                           1.23
- audio/audacious-plugins/Makefile                              1.34
- audio/audacious/Makefile                                      1.28
- audio/audacious/buildlink3.mk                                 1.20
- audio/audacity/Makefile                                       1.78
- audio/aumix-gtk/Makefile                                      1.18
- audio/bmp-esound/Makefile                                     1.30
- audio/bmp-flac/Makefile                                       1.13
- audio/bmp-mac/Makefile                                        1.19
- audio/bmp-musepack/Makefile                                   1.27
- audio/bmp-pulse/Makefile                                      1.17
- audio/bmp-scrobbler/Makefile                                  1.15
- audio/bmp/Makefile                                            1.49
- audio/bmp/buildlink3.mk                                       1.23
- audio/bmpx/Makefile                                           1.48
- audio/bsl/Makefile                                            1.17
- audio/buzztard/Makefile                                       1.21
- audio/easytag/Makefile                                        1.76
- audio/exaile/Makefile                                         1.34
- audio/gbemol/Makefile                                         1.18
- audio/gimmix/Makefile                                         1.24
- audio/gkrellm-volume/Makefile                                 1.41
- audio/glurp/Makefile                                          1.28
- audio/gmp3info/Makefile                                       1.40
- audio/gmpc-albumview/Makefile                                 1.12
- audio/gmpc-lastfm/Makefile                                    1.11
- audio/gmpc-lyrics/Makefile                                    1.12
- audio/gmpc-lyricwiki/Makefile                                 1.14
- audio/gmpc-magnatune/Makefile                                 1.16
- audio/gmpc-mdcover/Makefile                                   1.11
- audio/gmpc-tagedit/Makefile                                   1.12
- audio/gmpc/Makefile                                           1.53
- audio/gmpc/buildlink3.mk                                      1.17
- audio/gnome-vfs-cdda/Makefile                                 1.20
- audio/gqmpeg-devel/Makefile                                   1.39
- audio/gtick/Makefile                                          1.26
- audio/gtkpod/Makefile                                         1.46
- audio/hydrogen/Makefile                                       1.31
- audio/jack-rack/Makefile                                      1.25
- audio/libcanberra/Makefile                                    1.26
- audio/libcanberra/buildlink3.mk                               1.16
- audio/libgpod/Makefile                                        1.29
- audio/libgpod/buildlink3.mk                                   1.15
- audio/liblastfm/Makefile                                      1.22
- audio/liblastfm/buildlink3.mk                                 1.14
- audio/libvisual-bmp/Makefile                                  1.31
- audio/libvisual-plugins/Makefile                              1.35
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- sysutils/libnotify/buildlink3.mk                              1.20
- sysutils/nautilus-sendto/Makefile                             1.33
- sysutils/nautilus/Makefile                                    1.126
- sysutils/nautilus/buildlink3.mk                               1.51
- sysutils/notification-daemon/Makefile                         1.28
- sysutils/open-vm-tools/Makefile                               1.29
- sysutils/p5-Gnome2-VFS/Makefile                               1.15
- sysutils/p5-Gnome2-VFS/buildlink3.mk                          1.14
- sysutils/pcmanfm/Makefile                                     1.20
- sysutils/py-notify/Makefile                                   1.26
- sysutils/py-notify/buildlink3.mk                              1.20
- sysutils/rox/Makefile                                         1.64
- sysutils/screentest/Makefile                                  1.27
- sysutils/strigi/Makefile                                      1.23
- sysutils/virt-manager/Makefile                                1.4
- sysutils/xcdroast/Makefile                                    1.63
- sysutils/xfce4-appfinder/Makefile                             1.26
- sysutils/xfce4-battery-plugin/Makefile                        1.22
- sysutils/xfce4-battery-plugin/buildlink3.mk                   1.19
- sysutils/xfce4-cpugraph-plugin/Makefile                       1.25
- sysutils/xfce4-cpugraph-plugin/buildlink3.mk                  1.29
- sysutils/xfce4-diskperf-plugin/Makefile                       1.29
- sysutils/xfce4-diskperf-plugin/buildlink3.mk                  1.30
- sysutils/xfce4-fsguard-plugin/Makefile                        1.22
- sysutils/xfce4-fsguard-plugin/buildlink3.mk                   1.27
- sysutils/xfce4-genmon-plugin/Makefile                         1.15
- sysutils/xfce4-genmon-plugin/buildlink3.mk                    1.17
- sysutils/xfce4-netload-plugin/Makefile                        1.31
- sysutils/xfce4-netload-plugin/buildlink3.mk                   1.33
- sysutils/xfce4-quicklauncher-plugin/Makefile                  1.23
- sysutils/xfce4-quicklauncher-plugin/buildlink3.mk             1.29
- sysutils/xfce4-systemload-plugin/Makefile                     1.29
- sysutils/xfce4-systemload-plugin/buildlink3.mk                1.31
- sysutils/xfce4-thunar/Makefile                                1.28
- sysutils/xfce4-thunar/buildlink3.mk                           1.24
- sysutils/xfce4-volman/Makefile                                1.20
- sysutils/xfce4-xarchiver/Makefile                             1.19
- sysutils/xfce4-xarchiver/buildlink3.mk                        1.18
- sysutils/xfce4-xkb-plugin/Makefile                            1.27
- sysutils/xfce4-xkb-plugin/buildlink3.mk                       1.30
- textproc/FlightCrew/Makefile                                  1.16
- textproc/OdfConverter/Makefile                                1.20
- textproc/dikt/Makefile                                        1.10
- textproc/ebview/Makefile                                      1.17
- textproc/gnome-spell/Makefile                                 1.50
- textproc/gnome-spell/buildlink3.mk                            1.36
- textproc/gnome-subtitles/Makefile                             1.27
- textproc/gtkspell/Makefile                                    1.49
- textproc/gtkspell/buildlink3.mk                               1.26
- textproc/py-gdick/Makefile                                    1.32
- textproc/soprano/Makefile                                     1.29
- textproc/soprano/buildlink3.mk                                1.17
- textproc/xfce4-dict-plugin/Makefile                           1.15
- textproc/xfce4-dict-plugin/buildlink3.mk                      1.17
- time/cairo-clock/Makefile                                     1.27
- time/evolution-webcal/Makefile                                1.59
- time/gchore/Makefile                                          1.18
- time/gdeskcal/Makefile                                        1.44
- time/gnotime/Makefile                                         1.55
- time/gtodo-applet/Makefile                                    1.47
- time/gtodo/Makefile                                           1.43
- time/hamster-applet/Makefile                                  1.29
- time/ktimer/Makefile                                          1.6
- time/planner/Makefile                                         1.37
- time/rsibreak/Makefile                                        1.28
- time/xfce4-datetime-plugin/Makefile                           1.29
- time/xfce4-datetime-plugin/buildlink3.mk                      1.29
- time/xfce4-orage/Makefile                                     1.19
- time/xfce4-timer-plugin/Makefile                              1.15
- time/xfce4-timer-plugin/buildlink3.mk                         1.17
- wm/afterstep/Makefile                                         1.58
- wm/awesome/Makefile                                           1.31
- wm/bmpanel2/Makefile                                          1.10
- wm/compiz-fusion-plugins-extra/Makefile                       1.20
- wm/compiz-fusion-plugins-main/Makefile                        1.22
- wm/compiz-fusion-plugins-main/buildlink3.mk                   1.17
- wm/compiz/Makefile                                            1.52
- wm/compiz/buildlink3.mk                                       1.17
- wm/e16menuedit2/Makefile                                      1.35
- wm/fluxconf/Makefile                                          1.16
- wm/metacity/Makefile                                          1.103
- wm/metacity/buildlink3.mk                                     1.33
- wm/metisse/Makefile                                           1.27
- wm/openbox/Makefile                                           1.78
- wm/oroborox/Makefile                                          1.34
- wm/sawfish/Makefile                                           1.67
- wm/selectwm/Makefile                                          1.21
- wm/wbar/Makefile                                              1.11
- wm/wmakerconf/Makefile                                        1.57
- wm/xfce4-wm-themes/Makefile                                   1.31
- wm/xfce4-wm/Makefile                                          1.43
- www/amaya/Makefile                                            1.74
- www/ap22-dnssd/Makefile                                       1.18
- www/bluefish/Makefile                                         1.101
- www/browser-bookmarks-menu/Makefile                           1.33
- www/drivel/Makefile                                           1.65
- www/epiphany-extensions/Makefile                              1.76
- www/epiphany/Makefile                                         1.131
- www/epiphany/buildlink3.mk                                    1.50
- www/firefox/Makefile                                          1.112
- www/firefox10/Makefile                                        1.8
- www/firefox36/Makefile                                        1.18
- www/gtkhtml314/Makefile                                       1.52
- www/icedtea-web/Makefile                                      1.14
- www/kazehakase/Makefile                                       1.50
- www/kdewebdev4/Makefile                                       1.31
- www/libgtkhtml/Makefile                                       1.62
- www/libgtkhtml/buildlink3.mk                                  1.35
- www/libmediawiki/Makefile                                     1.5
- www/libmediawiki/buildlink3.mk                                1.5
- www/liferea-current/Makefile                                  1.34
- www/liferea/Makefile                                          1.108
- www/midori/Makefile                                           1.48
- www/netsurf/Makefile                                          1.21
- www/nspluginwrapper/Makefile                                  1.32
- www/seamonkey/Makefile                                        1.78
- www/webkit-gtk/Makefile                                       1.59
- www/webkit-gtk/buildlink3.mk                                  1.27
- www/webkit-gtk3/Makefile                                      1.14
- www/webkit-gtk3/buildlink3.mk                                 1.10
- x11/alacarte/Makefile                                         1.41
- x11/antiright/Makefile                                        1.21
- x11/avant-window-navigator/Makefile                           1.11
- x11/clisp-gtk2/Makefile                                       1.13
- x11/deskmenu/Makefile                                         1.15
- x11/devilspie/Makefile                                        1.52
- x11/eekboard/Makefile                                         1.17
- x11/eekboard/buildlink3.mk                                    1.15
- x11/fast-user-switch-applet/Makefile                          1.45
- x11/fltk13/Makefile                                           1.4
- x11/ftmenu/Makefile                                           1.17
- x11/gdm/Makefile                                              1.170
- x11/gnome-applets/Makefile                                    1.105
- x11/gnome-control-center/Makefile                             1.55
- x11/gnome-control-center/buildlink3.mk                        1.28
- x11/gnome-desktop-sharp/Makefile                              1.38
- x11/gnome-desktop-sharp/buildlink3.mk                         1.31
- x11/gnome-desktop/Makefile                                    1.117
- x11/gnome-desktop/buildlink3.mk                               1.43
- x11/gnome-desktop3/Makefile                                   1.4
- x11/gnome-desktop3/buildlink3.mk                              1.4
- x11/gnome-mag/Makefile                                        1.73
- x11/gnome-mag/buildlink3.mk                                   1.31
- x11/gnome-panel/Makefile                                      1.148
- x11/gnome-panel/buildlink3.mk                                 1.52
- x11/gnome-screensaver/Makefile                                1.58
- x11/gnome-session/Makefile                                    1.139
- x11/gnome-sharp/Makefile                                      1.36
- x11/gnome-sharp/buildlink3.mk                                 1.30
- x11/gnome-terminal/Makefile                                   1.46
- x11/gnome-themes-extras/Makefile                              1.58
- x11/gnome-themes/Makefile                                     1.89
- x11/gnopernicus/Makefile                                      1.78
- x11/grandr_applet/Makefile                                    1.27
- x11/gromit/Makefile                                           1.24
- x11/gselt/Makefile                                            1.12
- x11/gtk-sharp/Makefile                                        1.30
- x11/gtk2+extra/Makefile                                       1.20
- x11/gtk2+extra/buildlink3.mk                                  1.23
- x11/gtk2-chtheme/Makefile                                     1.24
- x11/gtk2-engines-bluecurve/Makefile                           1.31
- x11/gtk2-engines-murrine/Makefile                             1.17
- x11/gtk2-engines/Makefile                                     1.77
- x11/gtk2-engines/buildlink3.mk                                1.26
- x11/gtk2-theme-switch/Makefile                                1.32
- x11/gtk2/Makefile                                             1.236
- x11/gtk2/buildlink3.mk                                        1.56
- x11/gtk3/Makefile                                             1.27
- x11/gtk3/buildlink3.mk                                        1.8
- x11/gtkada/Makefile                                           1.11
- x11/gtkada/buildlink3.mk                                      1.10
- x11/gtkglarea2/Makefile                                       1.25
- x11/gtkglarea2/buildlink3.mk                                  1.20
- x11/gtkmm-utils/Makefile                                      1.12
- x11/gtkmm-utils/buildlink3.mk                                 1.14
- x11/gtkmm/Makefile                                            1.85
- x11/gtkmm/buildlink3.mk                                       1.32
- x11/gtksourceview-sharp2/Makefile                             1.33
- x11/gtksourceview/Makefile                                    1.64
- x11/gtksourceview/buildlink3.mk                               1.32
- x11/gtksourceview2/Makefile                                   1.33
- x11/gtksourceview2/buildlink3.mk                              1.19
- x11/gtksourceview3/Makefile                                   1.4
- x11/gtksourceview3/buildlink3.mk                              1.4
- x11/gtkterm2/Makefile                                         1.27
- x11/hot-babe/Makefile                                         1.33
- x11/kactivities/Makefile                                      1.6
- x11/kactivities/buildlink3.mk                                 1.6
- x11/kde-baseapps4/Makefile                                    1.8
- x11/kde-runtime4/Makefile                                     1.12
- x11/kde-runtime4/buildlink3.mk                                1.6
- x11/kde-workspace4/Makefile                                   1.11
- x11/kde-workspace4/buildlink3.mk                              1.6
- x11/kdelibs4/Makefile                                         1.47
- x11/kdelibs4/buildlink3.mk                                    1.22
- x11/konsole/Makefile                                          1.6
- x11/lablgtk/Makefile                                          1.68
- x11/lablgtk/buildlink3.mk                                     1.22
- x11/libX11/Makefile                                           1.28
- x11/libdesktop-agnostic/Makefile                              1.10
- x11/libdesktop-agnostic/buildlink3.mk                         1.7
- x11/libgnomekbd/Makefile                                      1.37
- x11/libunique/Makefile                                        1.21
- x11/libunique/buildlink3.mk                                   1.15
- x11/libunique3/Makefile                                       1.9
- x11/libxcb/buildlink3.mk                                      1.3
- x11/libxfce4gui/Makefile                                      1.41
- x11/libxfce4gui/buildlink3.mk                                 1.40
- x11/libxfce4menu/Makefile                                     1.11
- x11/libxfce4menu/buildlink3.mk                                1.13
- x11/libxfce4util/Makefile                                     1.27
- x11/matchbox-panel-manager/Makefile                           1.24
- x11/mlterm/Makefile                                           1.72
- x11/nucleo/Makefile                                           1.29
- x11/nucleo/buildlink3.mk                                      1.24
- x11/p5-Alien-wxWidgets/Makefile                               1.20
- x11/p5-Wx-Perl-ProcessStream/Makefile                         1.19
- x11/p5-Wx/Makefile                                            1.24
- x11/p5-Wx/buildlink3.mk                                       1.15
- x11/p5-gtk2/Makefile                                          1.49
- x11/p5-gtk2/buildlink3.mk                                     1.16
- x11/py-gnome2-desktop/Makefile                                1.67
- x11/py-gnome2-desktop/buildlink3.mk                           1.21
- x11/py-gnome2-extras/Makefile                                 1.57
- x11/py-gnome2-extras/buildlink3.mk                            1.32
- x11/py-gnome2/Makefile                                        1.86
- x11/py-gnome2/buildlink3.mk                                   1.41
- x11/py-gtk2/Makefile                                          1.87
- x11/py-gtk2/buildlink3.mk                                     1.40
- x11/py-gtksourceview/Makefile                                 1.24
- x11/py-kde4/Makefile                                          1.14
- x11/py-kiwi/Makefile                                          1.22
- x11/py-qt4-qscintilla/Makefile                                1.22
- x11/py-qt4-qscintilla/buildlink3.mk                           1.16
- x11/py-qt4/Makefile                                           1.54
- x11/py-vte/Makefile                                           1.11
- x11/py-vte/buildlink3.mk                                      1.11
- x11/py-wxWidgets/Makefile                                     1.39
- x11/py-wxWidgets/buildlink3.mk                                1.33
- x11/qt4-creator/Makefile                                      1.15
- x11/qt4-docs/Makefile                                         1.23
- x11/qt4-libs/Makefile                                         1.78
- x11/qt4-libs/buildlink3.mk                                    1.34
- x11/qt4-mng/Makefile                                          1.17
- x11/qt4-mysql/Makefile                                        1.26
- x11/qt4-pgsql/Makefile                                        1.34
- x11/qt4-qdbus/Makefile                                        1.28
- x11/qt4-qdbus/buildlink3.mk                                   1.17
- x11/qt4-qscintilla/Makefile                                   1.21
- x11/qt4-qscintilla/buildlink3.mk                              1.15
- x11/qt4-sqlite3/Makefile                                      1.29
- x11/qt4-tiff/Makefile                                         1.26
- x11/qt4-tools/Makefile                                        1.54
- x11/qt4-tools/buildlink3.mk                                   1.24
- x11/qwt-qt4/Makefile                                          1.20
- x11/qwt-qt4/buildlink3.mk                                     1.17
- x11/qwt6-qt4/Makefile                                         1.5
- x11/qwt6-qt4/buildlink3.mk                                    1.4
- x11/qwtplot3d-qt4/Makefile                                    1.15
- x11/qwtplot3d-qt4/buildlink3.mk                               1.15
- x11/rep-gtk2/Makefile                                         1.22
- x11/rox-session/Makefile                                      1.31
- x11/ruby-gnome2-gtk/Makefile                                  1.36
- x11/ruby-gnome2-gtk/buildlink3.mk                             1.21
- x11/ruby-gnome2-gtksourceview2/Makefile                       1.25
- x11/ruby-gnome2-vte/Makefile                                  1.28
- x11/ruby-wxruby/Makefile                                      1.8
- x11/sakura/Makefile                                           1.10
- x11/startup-notification/Makefile                             1.26
- x11/startup-notification010/Makefile                          1.3
- x11/tint2/Makefile                                            1.19
- x11/vte/Makefile                                              1.92
- x11/vte/buildlink3.mk                                         1.33
- x11/vte029/Makefile                                           1.4
- x11/vte029/buildlink3.mk                                      1.4
- x11/wxGTK24/Makefile                                          1.22
- x11/wxGTK24/buildlink3.mk                                     1.25
- x11/wxGTK26-contrib/Makefile                                  1.15
- x11/wxGTK26-contrib/buildlink3.mk                             1.18
- x11/wxGTK26/Makefile                                          1.18
- x11/wxGTK26/buildlink3.mk                                     1.17
- x11/wxGTK28-contrib/Makefile                                  1.18
- x11/wxGTK28-contrib/buildlink3.mk                             1.18
- x11/wxGTK28/Makefile                                          1.26
- x11/wxGTK28/buildlink3.mk                                     1.19
- x11/wxhaskell/Makefile                                        1.38
- x11/x11vnc/Makefile                                           1.43
- x11/xcb-util-image/Makefile                                   1.2
- x11/xcb-util-keysyms/Makefile                                 1.2
- x11/xcb-util-renderutil/Makefile                              1.2
- x11/xcb-util-wm/Makefile                                      1.2
- x11/xcb-util/Makefile                                         1.9
- x11/xcb-util036/Makefile                                      1.2
- x11/xdialog/Makefile                                          1.34
- x11/xfce4-clipman-plugin/Makefile                             1.27
- x11/xfce4-clipman-plugin/buildlink3.mk                        1.31
- x11/xfce4-desktop/Makefile                                    1.48
- x11/xfce4-exo/Makefile                                        1.23
- x11/xfce4-exo/buildlink3.mk                                   1.18
- x11/xfce4-eyes-plugin/Makefile                                1.15
- x11/xfce4-eyes-plugin/buildlink3.mk                           1.17
- x11/xfce4-gtk2-engine/Makefile                                1.24
- x11/xfce4-notes-plugin/Makefile                               1.29
- x11/xfce4-notes-plugin/buildlink3.mk                          1.33
- x11/xfce4-panel/Makefile                                      1.51
- x11/xfce4-panel/buildlink3.mk                                 1.38
- x11/xfce4-places-plugin/Makefile                              1.18
- x11/xfce4-places-plugin/buildlink3.mk                         1.20
- x11/xfce4-screenshooter-plugin/Makefile                       1.15
- x11/xfce4-screenshooter-plugin/buildlink3.mk                  1.17
- x11/xfce4-session/Makefile                                    1.39
- x11/xfce4-session/buildlink3.mk                               1.34
- x11/xfce4-settings/Makefile                                   1.18
- x11/xfce4-terminal/Makefile                                   1.25
- x11/xfce4-utils/Makefile                                      1.46
- x11/xlockmore/Makefile                                        1.98
- x11/xlsclients/Makefile                                       1.8
- x11/xpad/Makefile                                             1.41
- x11/xplanet/Makefile                                          1.59
- x11/xscreensaver-demo/Makefile                                1.47
- x11/xvidcap-gtk2/Makefile                                     1.23
- x11/xwininfo/Makefile                                         1.8
- x11/zenity/Makefile                                           1.71

-------------------------------------------------------------------
   Module Name:	pkgsrc
   Committed By:	tron
   Date:		Tue Oct  2 07:39:30 UTC 2012

   Modified Files:
   	pkgsrc/graphics/cairo: Makefile buildlink3.mk options.mk

   Log Message:
   The "cairo" package needs at least version 1.6 of the "libxcb" library.
   Set the minimum version accordingly. The package now uses "libxcb" from
   "pkgsrc" under Mac OS X Snow Leopard which fixes the build.

   Bump package revision because of the dependence change.


   To generate a diff of this commit:
   cvs rdiff -u -r1.104 -r1.105 pkgsrc/graphics/cairo/Makefile
   cvs rdiff -u -r1.42 -r1.43 pkgsrc/graphics/cairo/buildlink3.mk
   cvs rdiff -u -r1.7 -r1.8 pkgsrc/graphics/cairo/options.mk

-------------------------------------------------------------------
   Module Name:	pkgsrc
   Committed By:	tron
   Date:		Tue Oct  2 12:09:42 UTC 2012

   Modified Files:
   	pkgsrc/graphics/cairo: buildlink3.mk options.mk
   	pkgsrc/multimedia/vlc: Makefile
   	pkgsrc/multimedia/vlc10: Makefile
   	pkgsrc/multimedia/vlc2: Makefile
   	pkgsrc/multimedia/xine-lib: Makefile
   	pkgsrc/pkgtools/x11-links: Makefile
   	pkgsrc/x11/libX11: Makefile
   	pkgsrc/x11/libxcb: buildlink3.mk
   	pkgsrc/x11/startup-notification: Makefile
   	pkgsrc/x11/startup-notification010: Makefile
   	pkgsrc/x11/xcb-util: Makefile
   	pkgsrc/x11/xcb-util-image: Makefile
   	pkgsrc/x11/xcb-util-keysyms: Makefile
   	pkgsrc/x11/xcb-util-renderutil: Makefile
   	pkgsrc/x11/xcb-util-wm: Makefile
   	pkgsrc/x11/xcb-util036: Makefile
   	pkgsrc/x11/xlsclients: Makefile
   	pkgsrc/x11/xwininfo: Makefile

   Log Message:
   Increase the minium required version of the "x11/libxcb" package to 1.6.
   There are at least two packages ("graphics/cairo" and "x11/xlsclients")
   which require this. And we definitely want to avoid the situation where
   both the based and "pkgsrc" version get used at the same time.

   Bump all the packages which use the "x11/libxcb" package except the two
   which already require version 1.6. For those package remove the manually
   set minimum required version.

   Change discussed with Thomas Klausner in private e-mail.


   To generate a diff of this commit:
   cvs rdiff -u -r1.43 -r1.44 pkgsrc/graphics/cairo/buildlink3.mk
   cvs rdiff -u -r1.8 -r1.9 pkgsrc/graphics/cairo/options.mk
   cvs rdiff -u -r1.139 -r1.140 pkgsrc/multimedia/vlc/Makefile
   cvs rdiff -u -r1.20 -r1.21 pkgsrc/multimedia/vlc10/Makefile
   cvs rdiff -u -r1.12 -r1.13 pkgsrc/multimedia/vlc2/Makefile
   cvs rdiff -u -r1.105 -r1.106 pkgsrc/multimedia/xine-lib/Makefile
   cvs rdiff -u -r1.134 -r1.135 pkgsrc/pkgtools/x11-links/Makefile
   cvs rdiff -u -r1.27 -r1.28 pkgsrc/x11/libX11/Makefile
   cvs rdiff -u -r1.2 -r1.3 pkgsrc/x11/libxcb/buildlink3.mk
   cvs rdiff -u -r1.25 -r1.26 pkgsrc/x11/startup-notification/Makefile
   cvs rdiff -u -r1.2 -r1.3 pkgsrc/x11/startup-notification010/Makefile
   cvs rdiff -u -r1.8 -r1.9 pkgsrc/x11/xcb-util/Makefile
   cvs rdiff -u -r1.1 -r1.2 pkgsrc/x11/xcb-util-image/Makefile
   cvs rdiff -u -r1.1 -r1.2 pkgsrc/x11/xcb-util-keysyms/Makefile
   cvs rdiff -u -r1.1 -r1.2 pkgsrc/x11/xcb-util-renderutil/Makefile
   cvs rdiff -u -r1.1 -r1.2 pkgsrc/x11/xcb-util-wm/Makefile
   cvs rdiff -u -r1.1 -r1.2 pkgsrc/x11/xcb-util036/Makefile
   cvs rdiff -u -r1.7 -r1.8 pkgsrc/x11/xlsclients/Makefile
   cvs rdiff -u -r1.7 -r1.8 pkgsrc/x11/xwininfo/Makefile

-------------------------------------------------------------------
   Module Name:	pkgsrc
   Committed By:	tron
   Date:		Tue Oct  2 12:17:10 UTC 2012

   Modified Files:
   	pkgsrc/pkgtools/x11-links: Makefile buildlink3.mk

   Log Message:
   Increase version number to 0.71 instead of bumping the revision. Also
   increase the minimum required version to 0.71.


   To generate a diff of this commit:
   cvs rdiff -u -r1.135 -r1.136 pkgsrc/pkgtools/x11-links/Makefile
   cvs rdiff -u -r1.43 -r1.44 pkgsrc/pkgtools/x11-links/buildlink3.mk

-------------------------------------------------------------------
   Module Name:	pkgsrc
   Committed By:	tron
   Date:		Tue Oct  2 17:12:24 UTC 2012

   Modified Files:
   	pkgsrc/archivers/ark: Makefile
   	pkgsrc/archivers/file-roller: Makefile
   	pkgsrc/audio/albumplayer: Makefile
   	pkgsrc/audio/amarok: Makefile
   	pkgsrc/audio/amarok-kde3: Makefile
   	pkgsrc/audio/ario: Makefile
   	pkgsrc/audio/audacious: Makefile buildlink3.mk
   	pkgsrc/audio/audacious-plugins: Makefile
   	pkgsrc/audio/audacity: Makefile
   	pkgsrc/audio/aumix-gtk: Makefile
   	pkgsrc/audio/bmp: Makefile buildlink3.mk
   	pkgsrc/audio/bmp-esound: Makefile
   	pkgsrc/audio/bmp-flac: Makefile
   	pkgsrc/audio/bmp-mac: Makefile
   	pkgsrc/audio/bmp-musepack: Makefile
   	pkgsrc/audio/bmp-pulse: Makefile
   	pkgsrc/audio/bmp-scrobbler: Makefile
   	pkgsrc/audio/bmpx: Makefile
   	pkgsrc/audio/bsl: Makefile
   	pkgsrc/audio/buzztard: Makefile
   	pkgsrc/audio/easytag: Makefile
   	pkgsrc/audio/exaile: Makefile
   	pkgsrc/audio/gbemol: Makefile
   	pkgsrc/audio/gimmix: Makefile
   	pkgsrc/audio/gkrellm-volume: Makefile
   	pkgsrc/audio/glurp: Makefile
   	pkgsrc/audio/gmp3info: Makefile
   	pkgsrc/audio/gmpc: Makefile buildlink3.mk
   	pkgsrc/audio/gmpc-albumview: Makefile
   	pkgsrc/audio/gmpc-lastfm: Makefile
   	pkgsrc/audio/gmpc-lyrics: Makefile
   	pkgsrc/audio/gmpc-lyricwiki: Makefile
   	pkgsrc/audio/gmpc-magnatune: Makefile
   	pkgsrc/audio/gmpc-mdcover: Makefile
   	pkgsrc/audio/gmpc-tagedit: Makefile
   	pkgsrc/audio/gnome-vfs-cdda: Makefile
   	pkgsrc/audio/gqmpeg-devel: Makefile
   	pkgsrc/audio/gtick: Makefile
   	pkgsrc/audio/gtkpod: Makefile
   	pkgsrc/audio/hydrogen: Makefile
   	pkgsrc/audio/jack-rack: Makefile
   	pkgsrc/audio/libcanberra: Makefile buildlink3.mk
   	pkgsrc/audio/libgpod: Makefile buildlink3.mk
   	pkgsrc/audio/liblastfm: Makefile buildlink3.mk
   	pkgsrc/audio/libvisual-bmp: Makefile
   	pkgsrc/audio/libvisual-plugins: Makefile
   	pkgsrc/audio/libvisual0.2-plugins: Makefile
   	pkgsrc/audio/liteamp: Makefile
   	pkgsrc/audio/mp3diags: Makefile
   	pkgsrc/audio/mp3splt-gtk: Makefile
   	pkgsrc/audio/muse: Makefile
   	pkgsrc/audio/padevchooser: Makefile
   	pkgsrc/audio/paman: Makefile
   	pkgsrc/audio/paprefs: Makefile
   	pkgsrc/audio/pavucontrol: Makefile
   	pkgsrc/audio/pavumeter: Makefile
   	pkgsrc/audio/pulseaudio: Makefile
   	pkgsrc/audio/py-musique: Makefile
   	pkgsrc/audio/qjackctl: Makefile
   	pkgsrc/audio/rhythmbox: Makefile
   	pkgsrc/audio/sfxr: Makefile
   	pkgsrc/audio/snd: Makefile
   	pkgsrc/audio/solfege: Makefile
   	pkgsrc/audio/sonata: Makefile
   	pkgsrc/audio/sound-juicer: Makefile
   	pkgsrc/audio/streamtuner: Makefile buildlink3.mk
   	pkgsrc/audio/sweep: Makefile
   	pkgsrc/audio/terminatorx: Makefile
   	pkgsrc/audio/timidity: Makefile
   	pkgsrc/audio/xfce4-mixer: Makefile
   	pkgsrc/audio/xfce4-xmms-plugin: Makefile buildlink3.mk
   	pkgsrc/biology/chemtool: Makefile
   	pkgsrc/biology/gnome-chemistry-utils: Makefile
   	pkgsrc/biology/openbabel: Makefile
   	pkgsrc/cad/boolean: Makefile
   	pkgsrc/cad/geda: Makefile
   	pkgsrc/cad/gerbv: Makefile
   	pkgsrc/cad/gsmc: Makefile
   	pkgsrc/cad/gtk2-wcalc: Makefile
   	pkgsrc/cad/gtkwave: Makefile
   	pkgsrc/cad/gwave: Makefile
   	pkgsrc/cad/librecad: Makefile
   	pkgsrc/cad/openscad: Makefile
   	pkgsrc/cad/pcb: Makefile
   	pkgsrc/cad/transcalc: Makefile
   	pkgsrc/chat/empathy: Makefile buildlink3.mk
   	pkgsrc/chat/finch: Makefile
   	pkgsrc/chat/gajim: Makefile
   	pkgsrc/chat/gg2: Makefile
   	pkgsrc/chat/gnomeicu: Makefile
   	pkgsrc/chat/gossip: Makefile
   	pkgsrc/chat/konversation: Makefile
   	pkgsrc/chat/libpurple: Makefile buildlink3.mk
   	pkgsrc/chat/msn-pecan: Makefile
   	pkgsrc/chat/pidgin: Makefile buildlink3.mk
   	pkgsrc/chat/pidgin-facebookchat: Makefile
   	pkgsrc/chat/pidgin-icb: Makefile
   	pkgsrc/chat/pidgin-latex: Makefile
   	pkgsrc/chat/pidgin-libnotify: Makefile
   	pkgsrc/chat/pidgin-otr: Makefile
   	pkgsrc/chat/pidgin-sametime: Makefile
   	pkgsrc/chat/pidgin-silc: Makefile
   	pkgsrc/chat/psi: Makefile
   	pkgsrc/chat/spectrum: Makefile
   	pkgsrc/chat/telepathy-farsight: Makefile buildlink3.mk
   	pkgsrc/chat/telepathy-mission-control: Makefile buildlink3.mk
   	pkgsrc/chat/xchat: Makefile
   	pkgsrc/chat/xchat-systray-integration: Makefile
   	pkgsrc/comms/asterisk10: Makefile
   	pkgsrc/comms/asterisk18: Makefile
   	pkgsrc/comms/efax-gtk: Makefile
   	pkgsrc/comms/gnome-pilot: Makefile buildlink3.mk
   	pkgsrc/comms/jpilot: Makefile buildlink3.mk
   	pkgsrc/comms/jpilot-syncmal: Makefile
   	pkgsrc/comms/multisync-gui: Makefile
   	pkgsrc/converters/tex2rtf: Makefile
   	pkgsrc/databases/gourmet: Makefile
   	pkgsrc/databases/gq: Makefile
   	pkgsrc/databases/gramps: Makefile
   	pkgsrc/databases/gramps2: Makefile
   	pkgsrc/databases/gramps3: Makefile
   	pkgsrc/databases/gtkdbfeditor: Makefile
   	pkgsrc/databases/libgnomedb: Makefile buildlink3.mk
   	pkgsrc/databases/mergeant: Makefile
   	pkgsrc/databases/mysql-workbench: Makefile
   	pkgsrc/databases/p5-RRD-Simple: Makefile
   	pkgsrc/databases/pgadmin3: Makefile
   	pkgsrc/databases/rrdtool: Makefile buildlink3.mk
   	pkgsrc/databases/ruby-rrdtool: Makefile
   	pkgsrc/devel/GConf-ui: Makefile buildlink3.mk
   	pkgsrc/devel/SDL_Pango: Makefile buildlink3.mk
   	pkgsrc/devel/anjuta: Makefile
   	pkgsrc/devel/at-spi: Makefile buildlink3.mk
   	pkgsrc/devel/automoc4: Makefile
   	pkgsrc/devel/blib: Makefile buildlink3.mk
   	pkgsrc/devel/boa-constructor: Makefile
   	pkgsrc/devel/coconut: Makefile
   	pkgsrc/devel/compizconfig-backend-gconf: Makefile
   	pkgsrc/devel/cppcheck: Makefile
   	pkgsrc/devel/cutter: Makefile
   	pkgsrc/devel/dconf-editor: Makefile
   	pkgsrc/devel/devhelp: Makefile buildlink3.mk
   	pkgsrc/devel/distccmon-gnome: Makefile
   	pkgsrc/devel/distccmon-gtk: Makefile
   	pkgsrc/devel/eric4: Makefile
   	pkgsrc/devel/ethos: Makefile buildlink3.mk
   	pkgsrc/devel/frama-c: Makefile
   	pkgsrc/devel/gconfmm: Makefile buildlink3.mk
   	pkgsrc/devel/gdl: Makefile buildlink3.mk
   	pkgsrc/devel/geany: Makefile
   	pkgsrc/devel/glade: Makefile
   	pkgsrc/devel/glade3: Makefile buildlink3.mk
   	pkgsrc/devel/gmtk: Makefile buildlink3.mk
   	pkgsrc/devel/gnome-build: Makefile buildlink3.mk
   	pkgsrc/devel/gobject-introspection: Makefile
   	pkgsrc/devel/gps: Makefile
   	pkgsrc/devel/gst-plugins0.10-pango: Makefile
   	pkgsrc/devel/guile-gnome: Makefile
   	pkgsrc/devel/kdesdk4: Makefile
   	pkgsrc/devel/kdevelop4: Makefile
   	pkgsrc/devel/kdevplatform: Makefile
   	pkgsrc/devel/kdiff3: Makefile
   	pkgsrc/devel/libbonoboui: Makefile buildlink3.mk
   	pkgsrc/devel/libcompizconfig: Makefile
   	pkgsrc/devel/libdbusmenu-qt: Makefile buildlink3.mk
   	pkgsrc/devel/libextractor: Makefile
   	pkgsrc/devel/libgail-gnome: Makefile buildlink3.mk
   	pkgsrc/devel/libglade: Makefile buildlink3.mk
   	pkgsrc/devel/libglademm: Makefile buildlink3.mk
   	pkgsrc/devel/libgnome: Makefile buildlink3.mk
   	pkgsrc/devel/libgnomemm: Makefile buildlink3.mk
   	pkgsrc/devel/libgnomeui: Makefile buildlink3.mk
   	pkgsrc/devel/libgnomeuimm: Makefile buildlink3.mk
   	pkgsrc/devel/libgweather: Makefile buildlink3.mk
   	pkgsrc/devel/libpeas: Makefile buildlink3.mk
   	pkgsrc/devel/libsexy: Makefile buildlink3.mk
   	pkgsrc/devel/libsexymm: Makefile buildlink3.mk
   	pkgsrc/devel/libview: Makefile buildlink3.mk
   	pkgsrc/devel/libwnck: Makefile buildlink3.mk
   	pkgsrc/devel/mono-tools: Makefile
   	pkgsrc/devel/monodevelop: Makefile buildlink3.mk
   	pkgsrc/devel/monodevelop-database: Makefile
   	pkgsrc/devel/monotone-viz: Makefile
   	pkgsrc/devel/p5-Gnome2: Makefile
   	pkgsrc/devel/p5-Gtk2-GladeXML: Makefile
   	pkgsrc/devel/p5-Wx-Perl-DataWalker: Makefile
   	pkgsrc/devel/p5-pango: Makefile buildlink3.mk
   	pkgsrc/devel/pango: Makefile buildlink3.mk
   	pkgsrc/devel/pangomm: Makefile buildlink3.mk
   	pkgsrc/devel/py-compizconfig: Makefile
   	pkgsrc/devel/py-gobject: Makefile
   	pkgsrc/devel/py-gobject-shared: Makefile
   	pkgsrc/devel/py-gobject3: Makefile
   	pkgsrc/devel/qconf: Makefile
   	pkgsrc/devel/qjson: Makefile
   	pkgsrc/devel/qtscriptgenerator: Makefile buildlink3.mk
   	pkgsrc/devel/rapidsvn: Makefile
   	pkgsrc/devel/ruby-gnome2-pango: Makefile buildlink3.mk
   	pkgsrc/devel/scintilla: Makefile buildlink3.mk
   	pkgsrc/devel/swt: Makefile
   	pkgsrc/devel/valide: Makefile
   	pkgsrc/devel/xfce4-dev-tools: Makefile
   	pkgsrc/devel/xfconf: Makefile
   	pkgsrc/devel/xulrunner: Makefile buildlink3.mk
   	pkgsrc/devel/xulrunner10: Makefile buildlink3.mk
   	pkgsrc/devel/xulrunner192: Makefile buildlink3.mk
   	pkgsrc/editors/Sigil: Makefile
   	pkgsrc/editors/TeXmacs-devel: Makefile
   	pkgsrc/editors/abiword: Makefile
   	pkgsrc/editors/abiword-plugins: Makefile
   	pkgsrc/editors/beaver: Makefile
   	pkgsrc/editors/conglomerate: Makefile
   	pkgsrc/editors/dasher: Makefile
   	pkgsrc/editors/emacs: Makefile
   	pkgsrc/editors/emacs-snapshot: Makefile
   	pkgsrc/editors/emacs22: Makefile
   	pkgsrc/editors/emacs24: Makefile
   	pkgsrc/editors/gconf-editor: Makefile
   	pkgsrc/editors/gedit: Makefile buildlink3.mk
   	pkgsrc/editors/gedit-python: Makefile
   	pkgsrc/editors/gedit3: Makefile
   	pkgsrc/editors/gedit3-spell: Makefile
   	pkgsrc/editors/ghex: Makefile
   	pkgsrc/editors/gice: Makefile
   	pkgsrc/editors/gobby: Makefile
   	pkgsrc/editors/gtranslator: Makefile
   	pkgsrc/editors/gummi: Makefile
   	pkgsrc/editors/kate: Makefile
   	pkgsrc/editors/kile: Makefile
   	pkgsrc/editors/leafpad: Makefile
   	pkgsrc/editors/lyx: Makefile
   	pkgsrc/editors/medit: Makefile
   	pkgsrc/editors/mlview: Makefile
   	pkgsrc/editors/notecase: Makefile
   	pkgsrc/editors/p5-Padre: Makefile
   	pkgsrc/editors/p5-Wx-Scintilla: Makefile
   	pkgsrc/editors/poedit: Makefile
   	pkgsrc/editors/qgoogletranslator: Makefile
   	pkgsrc/editors/scite: Makefile
   	pkgsrc/editors/tea: Makefile
   	pkgsrc/editors/texmaker: Makefile
   	pkgsrc/editors/texworks: Makefile
   	pkgsrc/editors/tomboy: Makefile
   	pkgsrc/editors/vim-gtk2: Makefile
   	pkgsrc/editors/xfce4-mousepad: Makefile
   	pkgsrc/editors/xournal: Makefile
   	pkgsrc/emulators/blinkensim: Makefile
   	pkgsrc/emulators/bochs: Makefile
   	pkgsrc/emulators/e-uae: Makefile
   	pkgsrc/emulators/fuse: Makefile
   	pkgsrc/emulators/gambatte: Makefile
   	pkgsrc/emulators/gens: Makefile
   	pkgsrc/emulators/gpsim: Makefile
   	pkgsrc/emulators/gpsim-devel: Makefile buildlink3.mk
   	pkgsrc/emulators/gpsim-oscilloscope: Makefile
   	pkgsrc/emulators/gpsim-ptyusart: Makefile
   	pkgsrc/emulators/mame: Makefile
   	pkgsrc/emulators/mess: Makefile
   	pkgsrc/emulators/nestopia: Makefile
   	pkgsrc/emulators/snes9x-gtk: Makefile
   	pkgsrc/emulators/tme: Makefile
   	pkgsrc/emulators/uae: Makefile
   	pkgsrc/emulators/vice: Makefile
   	pkgsrc/emulators/xnp2: Makefile
   	pkgsrc/emulators/yabause: Makefile
   	pkgsrc/filesystems/fuse-pod: Makefile
   	pkgsrc/finance/gkrellm-stock: Makefile
   	pkgsrc/finance/gnucash: Makefile
   	pkgsrc/fonts/fntsample: Makefile
   	pkgsrc/fonts/fontforge: Makefile
   	pkgsrc/fonts/gbdfed: Makefile
   	pkgsrc/fonts/gucharmap: Makefile buildlink3.mk
   	pkgsrc/games/atomix: Makefile
   	pkgsrc/games/blinken: Makefile
   	pkgsrc/games/bluemoon: Makefile
   	pkgsrc/games/ccgo: Makefile
   	pkgsrc/games/crack-attack: Makefile
   	pkgsrc/games/crossfire-client-gtk2: Makefile
   	pkgsrc/games/dopewars: Makefile
   	pkgsrc/games/eboard: Makefile
   	pkgsrc/games/flightgear: Makefile
   	pkgsrc/games/freeciv-client: Makefile
   	pkgsrc/games/frozen-bubble: Makefile
   	pkgsrc/games/gamazons: Makefile
   	pkgsrc/games/gbrainy: Makefile
   	pkgsrc/games/gcompris: Makefile
   	pkgsrc/games/genecys-client: Makefile
   	pkgsrc/games/gnome-games: Makefile
   	pkgsrc/games/gnome-mastermind: Makefile
   	pkgsrc/games/grhino: Makefile
   	pkgsrc/games/gtetrinet: Makefile
   	pkgsrc/games/gtkballs: Makefile
   	pkgsrc/games/gtklevel9: Makefile
   	pkgsrc/games/gtkmagnetic: Makefile
   	pkgsrc/games/gturing: Makefile
   	pkgsrc/games/holtz: Makefile
   	pkgsrc/games/kanagram: Makefile
   	pkgsrc/games/kanatest: Makefile
   	pkgsrc/games/kdegames4: Makefile
   	pkgsrc/games/kdetoys4: Makefile
   	pkgsrc/games/khangman: Makefile
   	pkgsrc/games/kye: Makefile
   	pkgsrc/games/pioneers: Makefile
   	pkgsrc/games/pokerth: Makefile
   	pkgsrc/games/puzzles: Makefile
   	pkgsrc/games/pytraffic: Makefile
   	pkgsrc/games/quarry: Makefile
   	pkgsrc/games/rftg: Makefile
   	pkgsrc/games/scummvm-tools: Makefile
   	pkgsrc/games/simgear: Makefile
   	pkgsrc/games/sirius: Makefile
   	pkgsrc/games/teg: Makefile
   	pkgsrc/games/wesnoth: Makefile
   	pkgsrc/games/wxhugo: Makefile
   	pkgsrc/games/xcowsay: Makefile
   	pkgsrc/geography/emerillon: Makefile
   	pkgsrc/geography/geoclue: Makefile
   	pkgsrc/geography/libchamplain04: Makefile buildlink3.mk
   	pkgsrc/geography/libchamplain06: Makefile buildlink3.mk
   	pkgsrc/geography/libchamplain08: Makefile buildlink3.mk
   	pkgsrc/geography/mapserver: Makefile
   	pkgsrc/geography/merkaartor: Makefile
   	pkgsrc/geography/qgis: Makefile
   	pkgsrc/geography/qlandkartegt: Makefile
   	pkgsrc/geography/qlandkartem: Makefile
   	pkgsrc/geography/viking: Makefile
   	pkgsrc/graphics/aqsis: Makefile
   	pkgsrc/graphics/blinkenthemes: Makefile
   	pkgsrc/graphics/blinkentools: Makefile
   	pkgsrc/graphics/cairo: Makefile buildlink3.mk
   	pkgsrc/graphics/cairo-gobject: Makefile buildlink3.mk
   	pkgsrc/graphics/cairomm: Makefile buildlink3.mk
   	pkgsrc/graphics/camlimages: Makefile
   	pkgsrc/graphics/cheese: Makefile
   	pkgsrc/graphics/cinepaint: Makefile
   	pkgsrc/graphics/clutter: Makefile buildlink3.mk
   	pkgsrc/graphics/clutter-box2d: Makefile
   	pkgsrc/graphics/clutter-cairo: Makefile buildlink3.mk
   	pkgsrc/graphics/clutter-gtk: Makefile buildlink3.mk
   	pkgsrc/graphics/clutter-mx: Makefile buildlink3.mk
   	pkgsrc/graphics/clutter-qt: Makefile
   	pkgsrc/graphics/clutter08: Makefile buildlink3.mk
   	pkgsrc/graphics/clutter08-gtk: Makefile buildlink3.mk
   	pkgsrc/graphics/comix: Makefile
   	pkgsrc/graphics/darktable: Makefile
   	pkgsrc/graphics/denemo: Makefile
   	pkgsrc/graphics/dia: Makefile
   	pkgsrc/graphics/dia-python: Makefile
   	pkgsrc/graphics/digikam: Makefile
   	pkgsrc/graphics/djview4: Makefile
   	pkgsrc/graphics/eog: Makefile buildlink3.mk
   	pkgsrc/graphics/eog-plugins: Makefile
   	pkgsrc/graphics/eog-plugins-map: Makefile
   	pkgsrc/graphics/eog3: Makefile
   	pkgsrc/graphics/evas-svg: Makefile
   	pkgsrc/graphics/extrema: Makefile
   	pkgsrc/graphics/f-spot: Makefile
   	pkgsrc/graphics/fotoxx: Makefile
   	pkgsrc/graphics/frameworks: Makefile
   	pkgsrc/graphics/geeqie: Makefile
   	pkgsrc/graphics/gegl: Makefile buildlink3.mk
   	pkgsrc/graphics/gfract: Makefile
   	pkgsrc/graphics/gimmage: Makefile
   	pkgsrc/graphics/gimp: Makefile buildlink3.mk
   	pkgsrc/graphics/gimp-color-manager: Makefile
   	pkgsrc/graphics/gimp-exif-browser: Makefile
   	pkgsrc/graphics/gimp-fix-ca: Makefile
   	pkgsrc/graphics/gimp-liquid-rescale: Makefile
   	pkgsrc/graphics/gimp-rawphoto: Makefile
   	pkgsrc/graphics/gimp-refocus-it: Makefile
   	pkgsrc/graphics/gimp-resynthesizer: Makefile
   	pkgsrc/graphics/gimp-ufraw: Makefile
   	pkgsrc/graphics/gimp2-wideangle: Makefile
   	pkgsrc/graphics/giram: Makefile
   	pkgsrc/graphics/gliv: Makefile
   	pkgsrc/graphics/gmngview: Makefile
   	pkgsrc/graphics/gnome-icon-theme: Makefile
   	pkgsrc/graphics/gnuplot: Makefile
   	pkgsrc/graphics/goocanvas: Makefile buildlink3.mk
   	pkgsrc/graphics/goocanvasmm: Makefile
   	pkgsrc/graphics/gpicview: Makefile
   	pkgsrc/graphics/gqview: Makefile
   	pkgsrc/graphics/gqview-devel: Makefile
   	pkgsrc/graphics/graphviz: Makefile
   	pkgsrc/graphics/gst-plugins0.10-cairo: Makefile
   	pkgsrc/graphics/gthumb: Makefile
   	pkgsrc/graphics/gthumb3: Makefile
   	pkgsrc/graphics/gtkam: Makefile
   	pkgsrc/graphics/gtkglext: Makefile buildlink3.mk
   	pkgsrc/graphics/gtkimageview: Makefile
   	pkgsrc/graphics/guile-cairo: Makefile
   	pkgsrc/graphics/gwenview: Makefile
   	pkgsrc/graphics/hugin: Makefile
   	pkgsrc/graphics/inkscape: Makefile
   	pkgsrc/graphics/kamera: Makefile
   	pkgsrc/graphics/kcolorchooser: Makefile
   	pkgsrc/graphics/kdegraphics-mobipocket: Makefile
   	pkgsrc/graphics/kdegraphics-strigi-analyzer: Makefile
   	pkgsrc/graphics/kdegraphics-thumbnailers: Makefile
   	pkgsrc/graphics/kgamma: Makefile
   	pkgsrc/graphics/kipi-plugins: Makefile
   	pkgsrc/graphics/kipi-plugins-calendar-kde3: Makefile
   	pkgsrc/graphics/kipi-plugins-kde3: Makefile
   	pkgsrc/graphics/kolourpaint: Makefile
   	pkgsrc/graphics/koverartist: Makefile
   	pkgsrc/graphics/kruler: Makefile
   	pkgsrc/graphics/ksaneplugin: Makefile
   	pkgsrc/graphics/ksnapshot: Makefile
   	pkgsrc/graphics/libexif-gtk: Makefile buildlink3.mk
   	pkgsrc/graphics/libgdiplus: Makefile buildlink3.mk
   	pkgsrc/graphics/libgnomecanvas: Makefile buildlink3.mk
   	pkgsrc/graphics/libgnomecanvasmm: Makefile buildlink3.mk
   	pkgsrc/graphics/libkdcraw: Makefile buildlink3.mk
   	pkgsrc/graphics/libkexiv2: Makefile
   	pkgsrc/graphics/libkface: Makefile buildlink3.mk
   	pkgsrc/graphics/libkipi: Makefile buildlink3.mk
   	pkgsrc/graphics/libksane: Makefile buildlink3.mk
   	pkgsrc/graphics/librsvg: Makefile buildlink3.mk
   	pkgsrc/graphics/libscigraphica: Makefile buildlink3.mk
   	pkgsrc/graphics/nvtv: Makefile
   	pkgsrc/graphics/opencv: Makefile
   	pkgsrc/graphics/osg: Makefile buildlink3.mk
   	pkgsrc/graphics/p5-Gnome2-Canvas: Makefile buildlink3.mk
   	pkgsrc/graphics/p5-cairo: Makefile buildlink3.mk
   	pkgsrc/graphics/p5-clutter: Makefile
   	pkgsrc/graphics/py-cairo: Makefile buildlink3.mk
   	pkgsrc/graphics/py-clutter: Makefile buildlink3.mk
   	pkgsrc/graphics/py-clutter-gtk: Makefile
   	pkgsrc/graphics/py-goocanvas: Makefile
   	pkgsrc/graphics/py-gtkglext: Makefile
   	pkgsrc/graphics/py-matplotlib-gtk2: Makefile
   	pkgsrc/graphics/qimageblitz: Makefile buildlink3.mk
   	pkgsrc/graphics/qiv: Makefile
   	pkgsrc/graphics/qiviewer: Makefile
   	pkgsrc/graphics/ristretto: Makefile
   	pkgsrc/graphics/ruby-clutter-cairo: Makefile
   	pkgsrc/graphics/ruby-clutter-core: Makefile
   	pkgsrc/graphics/ruby-clutter-gtk: Makefile
   	pkgsrc/graphics/ruby-gnome2-goocanvas: Makefile
   	pkgsrc/graphics/ruby-gnome2-rsvg: Makefile
   	pkgsrc/graphics/ruby-rcairo: Makefile buildlink3.mk
   	pkgsrc/graphics/sane-frontends: Makefile
   	pkgsrc/graphics/scidavis: Makefile
   	pkgsrc/graphics/scigraphica: Makefile
   	pkgsrc/graphics/shotwell: Makefile
   	pkgsrc/graphics/shotwell-gtk3: Makefile
   	pkgsrc/graphics/svgpart: Makefile
   	pkgsrc/graphics/tuxpaint: Makefile
   	pkgsrc/graphics/ucview: Makefile
   	pkgsrc/graphics/unicap: Makefile
   	pkgsrc/graphics/veusz: Makefile
   	pkgsrc/graphics/viewnior: Makefile
   	pkgsrc/graphics/wxsvg: Makefile buildlink3.mk
   	pkgsrc/graphics/xfce4-icon-theme: Makefile buildlink3.mk
   	pkgsrc/graphics/xsane: Makefile
   	pkgsrc/graphics/xzgv: Makefile
   	pkgsrc/graphics/zphoto: Makefile
   	pkgsrc/ham/gmfsk: Makefile
   	pkgsrc/ham/gnuradio-examples: Makefile
   	pkgsrc/ham/gnuradio-radio-astronomy: Makefile
   	pkgsrc/ham/gnuradio-wxgui: Makefile buildlink3.mk
   	pkgsrc/ham/gpredict: Makefile
   	pkgsrc/ham/grig: Makefile
   	pkgsrc/ham/xdx: Makefile
   	pkgsrc/ham/xlog: Makefile
   	pkgsrc/inputmethod/gcin: Makefile
   	pkgsrc/inputmethod/gtk-im-libthai: Makefile
   	pkgsrc/inputmethod/ibus: Makefile
   	pkgsrc/inputmethod/ibus-chewing: Makefile
   	pkgsrc/inputmethod/ibus-handwrite: Makefile
   	pkgsrc/inputmethod/ibus-input-pad: Makefile
   	pkgsrc/inputmethod/ibus-m17n: Makefile
   	pkgsrc/inputmethod/ibus-qt: Makefile
   	pkgsrc/inputmethod/ibus-skk: Makefile
   	pkgsrc/inputmethod/ibus-t9: Makefile
   	pkgsrc/inputmethod/ibus-unikey: Makefile
   	pkgsrc/inputmethod/im-ja: Makefile
   	pkgsrc/inputmethod/imhangul: Makefile
   	pkgsrc/inputmethod/imhangul_status_applet: Makefile
   	pkgsrc/inputmethod/input-pad: Makefile buildlink3.mk
   	pkgsrc/inputmethod/kasumi: Makefile
   	pkgsrc/inputmethod/kimera: Makefile
   	pkgsrc/inputmethod/kimera-tomoe-gtk: Makefile
   	pkgsrc/inputmethod/m17n-im-config: Makefile buildlink3.mk
   	pkgsrc/inputmethod/nabi: Makefile
   	pkgsrc/inputmethod/novel-pinyin: Makefile
   	pkgsrc/inputmethod/py-input-pad: Makefile
   	pkgsrc/inputmethod/scim: Makefile buildlink3.mk
   	pkgsrc/inputmethod/scim-anthy: Makefile
   	pkgsrc/inputmethod/scim-array: Makefile
   	pkgsrc/inputmethod/scim-canna: Makefile
   	pkgsrc/inputmethod/scim-ccinput: Makefile
   	pkgsrc/inputmethod/scim-chewing: Makefile
   	pkgsrc/inputmethod/scim-fcitx: Makefile
   	pkgsrc/inputmethod/scim-hangul: Makefile
   	pkgsrc/inputmethod/scim-input-pad: Makefile
   	pkgsrc/inputmethod/scim-m17n: Makefile
   	pkgsrc/inputmethod/scim-pinyin: Makefile
   	pkgsrc/inputmethod/scim-prime: Makefile
   	pkgsrc/inputmethod/scim-python: Makefile
   	pkgsrc/inputmethod/scim-sinhala: Makefile
   	pkgsrc/inputmethod/scim-skk: Makefile
   	pkgsrc/inputmethod/scim-tables: Makefile
   	pkgsrc/inputmethod/scim-thai: Makefile
   	pkgsrc/inputmethod/scim-tomoe: Makefile
   	pkgsrc/inputmethod/scim-uim: Makefile
   	pkgsrc/inputmethod/scim-unikey: Makefile
   	pkgsrc/inputmethod/tomoe-gtk: Makefile buildlink3.mk
   	pkgsrc/inputmethod/uim: Makefile
   	pkgsrc/inputmethod/uim-tomoe-gtk: Makefile
   	pkgsrc/lang/basic256: Makefile
   	pkgsrc/lang/classpath-gui: Makefile
   	pkgsrc/lang/clisp: Makefile
   	pkgsrc/lang/coq: Makefile
   	pkgsrc/lang/kaffe-x11: Makefile
   	pkgsrc/lang/racket: Makefile
   	pkgsrc/lang/sablevm-classpath-gui: Makefile
   	pkgsrc/lang/smalltalk: Makefile
   	pkgsrc/lang/squeak-vm: Makefile
   	pkgsrc/mail/akonadi: Makefile buildlink3.mk
   	pkgsrc/mail/balsa: Makefile
   	pkgsrc/mail/claws-mail: Makefile
   	pkgsrc/mail/claws-mail-archive: Makefile
   	pkgsrc/mail/claws-mail-attachwarner: Makefile
   	pkgsrc/mail/claws-mail-attremover: Makefile
   	pkgsrc/mail/claws-mail-cachesaver: Makefile
   	pkgsrc/mail/claws-mail-dillo-viewer: Makefile
   	pkgsrc/mail/claws-mail-fetchinfo: Makefile
   	pkgsrc/mail/claws-mail-gtkhtml: Makefile
   	pkgsrc/mail/claws-mail-mailmbox: Makefile
   	pkgsrc/mail/claws-mail-newmail: Makefile
   	pkgsrc/mail/claws-mail-notification: Makefile
   	pkgsrc/mail/claws-mail-pgpcore: Makefile buildlink3.mk
   	pkgsrc/mail/claws-mail-pgpinline: Makefile
   	pkgsrc/mail/claws-mail-pgpmime: Makefile
   	pkgsrc/mail/claws-mail-rssyl: Makefile
   	pkgsrc/mail/claws-mail-smime: Makefile
   	pkgsrc/mail/claws-mail-spamreport: Makefile
   	pkgsrc/mail/claws-mail-synce: Makefile
   	pkgsrc/mail/claws-mail-tnef: Makefile
   	pkgsrc/mail/claws-mail-vcalendar: Makefile
   	pkgsrc/mail/evolution: Makefile buildlink3.mk
   	pkgsrc/mail/evolution-data-server: Makefile
   	pkgsrc/mail/evolution-exchange: Makefile
   	pkgsrc/mail/mail-notification: Makefile
   	pkgsrc/mail/msmtp: Makefile
   	pkgsrc/mail/netbiff: Makefile
   	pkgsrc/mail/sylpheed: Makefile
   	pkgsrc/mail/thunderbird: Makefile
   	pkgsrc/mail/thunderbird10: Makefile
   	pkgsrc/math/R: Makefile
   	pkgsrc/math/analitza: Makefile buildlink3.mk
   	pkgsrc/math/calcoo: Makefile
   	pkgsrc/math/cantor: Makefile
   	pkgsrc/math/cgal: Makefile buildlink3.mk
   	pkgsrc/math/euler: Makefile
   	pkgsrc/math/extcalc: Makefile
   	pkgsrc/math/fityk: Makefile
   	pkgsrc/math/galculator: Makefile
   	pkgsrc/math/gcalctool: Makefile
   	pkgsrc/math/gcalctool-gtk3: Makefile
   	pkgsrc/math/genius: Makefile
   	pkgsrc/math/gnumeric: Makefile
   	pkgsrc/math/gnumeric110: Makefile
   	pkgsrc/math/grpn: Makefile
   	pkgsrc/math/gtklife: Makefile
   	pkgsrc/math/kalgebra: Makefile
   	pkgsrc/math/kcalc: Makefile
   	pkgsrc/math/pspp: Makefile
   	pkgsrc/math/qalculate-bases: Makefile
   	pkgsrc/math/qalculate-currency: Makefile
   	pkgsrc/math/qalculate-gtk: Makefile
   	pkgsrc/math/qalculate-units: Makefile
   	pkgsrc/math/speedcrunch: Makefile
   	pkgsrc/math/wxMaxima: Makefile
   	pkgsrc/meta-pkgs/xfce4: Makefile buildlink3.mk
   	pkgsrc/meta-pkgs/xfce4-extras: Makefile
   	pkgsrc/misc/attica: Makefile buildlink3.mk
   	pkgsrc/misc/blccc: Makefile
   	pkgsrc/misc/calibre: Makefile
   	pkgsrc/misc/celestia: Makefile
   	pkgsrc/misc/deskbar-applet: Makefile
   	pkgsrc/misc/fbreader: Makefile
   	pkgsrc/misc/gelemental: Makefile
   	pkgsrc/misc/gkrellm-flynn: Makefile
   	pkgsrc/misc/gkrellm-launch: Makefile
   	pkgsrc/misc/gkrellm-leds: Makefile
   	pkgsrc/misc/gkrellm-moon: Makefile
   	pkgsrc/misc/gkrellm-weather: Makefile
   	pkgsrc/misc/gnome-utils: Makefile
   	pkgsrc/misc/goffice0.6: Makefile buildlink3.mk
   	pkgsrc/misc/goffice0.8: Makefile buildlink3.mk
   	pkgsrc/misc/gok: Makefile
   	pkgsrc/misc/gperiodic: Makefile
   	pkgsrc/misc/gwaei: Makefile
   	pkgsrc/misc/kaccessible: Makefile
   	pkgsrc/misc/kalzium: Makefile
   	pkgsrc/misc/kanjipad: Makefile
   	pkgsrc/misc/kbruch: Makefile
   	pkgsrc/misc/kcharselect: Makefile
   	pkgsrc/misc/kchmviewer: Makefile
   	pkgsrc/misc/kde-wallpapers4: Makefile
   	pkgsrc/misc/kdeadmin4: Makefile
   	pkgsrc/misc/kdeartwork4: Makefile
   	pkgsrc/misc/kdepim-runtime4: Makefile
   	pkgsrc/misc/kdepim4: Makefile
   	pkgsrc/misc/kdepimlibs4: Makefile buildlink3.mk
   	pkgsrc/misc/kdeplasma-addons4: Makefile
   	pkgsrc/misc/kgeography: Makefile
   	pkgsrc/misc/kig: Makefile
   	pkgsrc/misc/kiten: Makefile
   	pkgsrc/misc/klettres: Makefile
   	pkgsrc/misc/kmag: Makefile
   	pkgsrc/misc/kmousetool: Makefile
   	pkgsrc/misc/kmouth: Makefile
   	pkgsrc/misc/kremotecontrol: Makefile
   	pkgsrc/misc/kstars: Makefile
   	pkgsrc/misc/ktouch: Makefile
   	pkgsrc/misc/kturtle: Makefile
   	pkgsrc/misc/kwordquiz: Makefile
   	pkgsrc/misc/libkdeedu: Makefile buildlink3.mk
   	pkgsrc/misc/libkgeomap: Makefile buildlink3.mk
   	pkgsrc/misc/libkvkontakte: Makefile buildlink3.mk
   	pkgsrc/misc/libreoffice: Makefile
   	pkgsrc/misc/marble: Makefile buildlink3.mk
   	pkgsrc/misc/mousetweaks: Makefile
   	pkgsrc/misc/openoffice3: Makefile
   	pkgsrc/misc/parley: Makefile
   	pkgsrc/misc/rocs: Makefile
   	pkgsrc/misc/stellarium: Makefile
   	pkgsrc/misc/step: Makefile
   	pkgsrc/misc/superkaramba: Makefile
   	pkgsrc/misc/sweeper: Makefile
   	pkgsrc/misc/tellico: Makefile
   	pkgsrc/misc/usbprog: Makefile
   	pkgsrc/misc/vym: Makefile
   	pkgsrc/misc/xchm: Makefile
   	pkgsrc/misc/xfce4-weather-plugin: Makefile buildlink3.mk
   	pkgsrc/misc/yelp: Makefile
   	pkgsrc/misc/yelp3: Makefile
   	pkgsrc/misc/zyGrib: Makefile
   	pkgsrc/multimedia/avidemux: Makefile
   	pkgsrc/multimedia/clutter-gst: Makefile buildlink3.mk
   	pkgsrc/multimedia/clutter08-gst: Makefile buildlink3.mk
   	pkgsrc/multimedia/dvdstyler: Makefile
   	pkgsrc/multimedia/farsight2: Makefile buildlink3.mk
   	pkgsrc/multimedia/gecko-mediaplayer: Makefile
   	pkgsrc/multimedia/gmencoder: Makefile
   	pkgsrc/multimedia/gmplayer: Makefile
   	pkgsrc/multimedia/gnash: Makefile
   	pkgsrc/multimedia/gnome-media: Makefile buildlink3.mk
   	pkgsrc/multimedia/gnome-mplayer: Makefile
   	pkgsrc/multimedia/gopchop: Makefile
   	pkgsrc/multimedia/gst123: Makefile
   	pkgsrc/multimedia/gxine: Makefile
   	pkgsrc/multimedia/handbrake: Makefile
   	pkgsrc/multimedia/hwdecode-demos: Makefile
   	pkgsrc/multimedia/kdemultimedia4: Makefile
   	pkgsrc/multimedia/kdenlive: Makefile
   	pkgsrc/multimedia/kmplayer: Makefile
   	pkgsrc/multimedia/kmplayer-kde3: Makefile
   	pkgsrc/multimedia/mkvtoolnix: Makefile
   	pkgsrc/multimedia/mkvtoolnix-old: Makefile
   	pkgsrc/multimedia/mlt: Makefile
   	pkgsrc/multimedia/mplayer-plugin: Makefile
   	pkgsrc/multimedia/ogle_gui: Makefile
   	pkgsrc/multimedia/phonon: Makefile buildlink3.mk
   	pkgsrc/multimedia/phonon-backend-gstreamer: Makefile
   	pkgsrc/multimedia/pitivi: Makefile
   	pkgsrc/multimedia/pspvc: Makefile
   	pkgsrc/multimedia/py-clutter-gst: Makefile
   	pkgsrc/multimedia/ruby-clutter-gst: Makefile
   	pkgsrc/multimedia/subtitleeditor: Makefile
   	pkgsrc/multimedia/swfdec: Makefile buildlink3.mk
   	pkgsrc/multimedia/swfdec-gnome: Makefile
   	pkgsrc/multimedia/swfdec-mozilla: Makefile
   	pkgsrc/multimedia/totem: Makefile buildlink3.mk
   	pkgsrc/multimedia/totem-browser-plugin: Makefile
   	pkgsrc/multimedia/totem-nautilus: Makefile
   	pkgsrc/multimedia/totem-pl-parser: Makefile buildlink3.mk
   	pkgsrc/multimedia/vlc: Makefile
   	pkgsrc/multimedia/vlc08: Makefile
   	pkgsrc/multimedia/vlc10: Makefile
   	pkgsrc/multimedia/vlc2: Makefile
   	pkgsrc/multimedia/xfce4-mpc-plugin: Makefile buildlink3.mk
   	pkgsrc/multimedia/xfmedia: Makefile
   	pkgsrc/multimedia/xvid4conf: Makefile
   	pkgsrc/net/Transmission-gui: Makefile
   	pkgsrc/net/amule: Makefile
   	pkgsrc/net/avahi: Makefile buildlink3.mk
   	pkgsrc/net/bug-buddy: Makefile
   	pkgsrc/net/cacti: Makefile
   	pkgsrc/net/choqok: Makefile
   	pkgsrc/net/dc_gui2: Makefile
   	pkgsrc/net/dcsharp: Makefile
   	pkgsrc/net/dhcpcd-gtk: Makefile
   	pkgsrc/net/ed2k-gtk-gui: Makefile
   	pkgsrc/net/ekiga: Makefile
   	pkgsrc/net/etherape: Makefile
   	pkgsrc/net/ettercap: Makefile
   	pkgsrc/net/filezilla: Makefile
   	pkgsrc/net/gitso: Makefile
   	pkgsrc/net/gkrellm-multiping: Makefile
   	pkgsrc/net/gkrellm-snmp: Makefile
   	pkgsrc/net/gkrellm-wireless: Makefile
   	pkgsrc/net/gnome-netstatus: Makefile
   	pkgsrc/net/gnome-nettool: Makefile
   	pkgsrc/net/gnome-vfs-smb: Makefile
   	pkgsrc/net/gssdp: Makefile
   	pkgsrc/net/gtk-gnutella: Makefile
   	pkgsrc/net/gtk-vnc: Makefile buildlink3.mk
   	pkgsrc/net/gupnp-tools: Makefile
   	pkgsrc/net/jigdo: Makefile
   	pkgsrc/net/kdenetwork4: Makefile
   	pkgsrc/net/kmldonkey: Makefile
   	pkgsrc/net/ktorrent: Makefile
   	pkgsrc/net/libdmapsharing: Makefile
   	pkgsrc/net/libktorrent: Makefile buildlink3.mk
   	pkgsrc/net/logjam: Makefile
   	pkgsrc/net/mbrowse: Makefile
   	pkgsrc/net/mldonkey-gui: Makefile
   	pkgsrc/net/monsoon: Makefile
   	pkgsrc/net/mtr: Makefile
   	pkgsrc/net/nagstamon: Makefile
   	pkgsrc/net/nfdump: Makefile
   	pkgsrc/net/nmap: Makefile
   	pkgsrc/net/ntop: Makefile
   	pkgsrc/net/remmina: Makefile
   	pkgsrc/net/remmina-plugins: Makefile
   	pkgsrc/net/tsclient: Makefile
   	pkgsrc/net/twitux: Makefile
   	pkgsrc/net/unison: Makefile
   	pkgsrc/net/unison2.32: Makefile
   	pkgsrc/net/urlgfe: Makefile
   	pkgsrc/net/vinagre: Makefile
   	pkgsrc/net/vino: Makefile
   	pkgsrc/net/wireshark: Makefile
   	pkgsrc/net/wistumbler2-gtk: Makefile
   	pkgsrc/net/xfce4-wavelan-plugin: Makefile buildlink3.mk
   	pkgsrc/net/xymon: Makefile
   	pkgsrc/news/pan: Makefile
   	pkgsrc/parallel/ganglia-monitor-core: Makefile
   	pkgsrc/parallel/hwloc: Makefile
   	pkgsrc/pkgtools/gnome-packagekit: Makefile
   	pkgsrc/pkgtools/packagekit: Makefile
   	pkgsrc/print/abcm2ps: Makefile
   	pkgsrc/print/advi: Makefile
   	pkgsrc/print/diffpdf: Makefile
   	pkgsrc/print/epdfview: Makefile
   	pkgsrc/print/evince: Makefile buildlink3.mk
   	pkgsrc/print/evince-nautilus: Makefile
   	pkgsrc/print/evince3: Makefile buildlink3.mk
   	pkgsrc/print/glabels: Makefile
   	pkgsrc/print/gtklp: Makefile
   	pkgsrc/print/gutenprint-lib: Makefile
   	pkgsrc/print/hplip: Makefile
   	pkgsrc/print/imposter: Makefile
   	pkgsrc/print/kbibtex: Makefile
   	pkgsrc/print/libgnomeprint: Makefile buildlink3.mk
   	pkgsrc/print/libgnomeprintui: Makefile buildlink3.mk
   	pkgsrc/print/libgxps: Makefile buildlink3.mk
   	pkgsrc/print/lilypond: Makefile
   	pkgsrc/print/okular: Makefile
   	pkgsrc/print/paps: Makefile
   	pkgsrc/print/poppler-glib: Makefile buildlink3.mk
   	pkgsrc/print/poppler-qt4: Makefile
   	pkgsrc/print/printer-applet: Makefile
   	pkgsrc/print/ruby-gnome2-poppler: Makefile
   	pkgsrc/print/scribus: Makefile
   	pkgsrc/print/scribus-qt4: Makefile
   	pkgsrc/print/xfce4-print: Makefile buildlink3.mk
   	pkgsrc/security/MyPasswordSafe: Makefile
   	pkgsrc/security/fprint-demo: Makefile
   	pkgsrc/security/fwbuilder: Makefile
   	pkgsrc/security/gnome-keyring: Makefile buildlink3.mk
   	pkgsrc/security/gnome-keyring-manager: Makefile
   	pkgsrc/security/gpa: Makefile
   	pkgsrc/security/gpass: Makefile
   	pkgsrc/security/honeyd: Makefile
   	pkgsrc/security/hydra: Makefile
   	pkgsrc/security/keepassx: Makefile
   	pkgsrc/security/kgpg: Makefile
   	pkgsrc/security/ksecrets: Makefile
   	pkgsrc/security/kwallet: Makefile
   	pkgsrc/security/libfwbuilder: Makefile buildlink3.mk
   	pkgsrc/security/openvas-client: Makefile
   	pkgsrc/security/pinentry-gtk2: Makefile
   	pkgsrc/security/pinentry-qt4: Makefile
   	pkgsrc/security/policykit-gnome: Makefile
   	pkgsrc/security/polkit-qt: Makefile buildlink3.mk
   	pkgsrc/security/putty: Makefile
   	pkgsrc/security/qca2: Makefile buildlink3.mk
   	pkgsrc/security/qca2-gnupg: Makefile
   	pkgsrc/security/qca2-ossl: Makefile
   	pkgsrc/security/qoauth: Makefile buildlink3.mk
   	pkgsrc/security/seahorse: Makefile buildlink3.mk
   	pkgsrc/security/seahorse-plugins: Makefile
   	pkgsrc/sysutils/amtterm: Makefile
   	pkgsrc/sysutils/bacula-qt-console: Makefile
   	pkgsrc/sysutils/bacula-tray-monitor: Makefile
   	pkgsrc/sysutils/bacula-wx-console: Makefile
   	pkgsrc/sysutils/brasero: Makefile buildlink3.mk
   	pkgsrc/sysutils/brasero-nautilus: Makefile
   	pkgsrc/sysutils/btpin-qt: Makefile
   	pkgsrc/sysutils/bubblemon: Makefile
   	pkgsrc/sysutils/conky: Makefile
   	pkgsrc/sysutils/dvdisaster: Makefile
   	pkgsrc/sysutils/filelight: Makefile
   	pkgsrc/sysutils/gcdmaster: Makefile
   	pkgsrc/sysutils/gkrellm: Makefile buildlink3.mk
   	pkgsrc/sysutils/gkrellm-est: Makefile
   	pkgsrc/sysutils/gnome-commander: Makefile
   	pkgsrc/sysutils/gnome-device-manager: Makefile
   	pkgsrc/sysutils/gnome-mount: Makefile buildlink3.mk
   	pkgsrc/sysutils/gnome-nds-thumbnailer: Makefile
   	pkgsrc/sysutils/gnome-pkgview: Makefile
   	pkgsrc/sysutils/gnome-power-manager: Makefile
   	pkgsrc/sysutils/gnome-settings-daemon: Makefile
   	pkgsrc/sysutils/gnome-system-monitor: Makefile
   	pkgsrc/sysutils/gnome-system-tools: Makefile
   	pkgsrc/sysutils/gnome-vfs: Makefile buildlink3.mk
   	pkgsrc/sysutils/gnome-vfs-monikers: Makefile
   	pkgsrc/sysutils/gnome-vfsmm: Makefile buildlink3.mk
   	pkgsrc/sysutils/gnome-volume-manager: Makefile
   	pkgsrc/sysutils/grun: Makefile
   	pkgsrc/sysutils/gst-plugins0.10-gnomevfs: Makefile
   	pkgsrc/sysutils/gtk-send-pr: Makefile
   	pkgsrc/sysutils/gvfs: Makefile
   	pkgsrc/sysutils/k3b: Makefile
   	pkgsrc/sysutils/k4dirstat: Makefile
   	pkgsrc/sysutils/kdf: Makefile
   	pkgsrc/sysutils/kfloppy: Makefile
   	pkgsrc/sysutils/libgksu: Makefile buildlink3.mk
   	pkgsrc/sysutils/libnotify: Makefile buildlink3.mk
   	pkgsrc/sysutils/nautilus: Makefile buildlink3.mk
   	pkgsrc/sysutils/nautilus-sendto: Makefile
   	pkgsrc/sysutils/notification-daemon: Makefile
   	pkgsrc/sysutils/open-vm-tools: Makefile
   	pkgsrc/sysutils/p5-Gnome2-VFS: Makefile buildlink3.mk
   	pkgsrc/sysutils/pcmanfm: Makefile
   	pkgsrc/sysutils/py-notify: Makefile buildlink3.mk
   	pkgsrc/sysutils/rox: Makefile
   	pkgsrc/sysutils/screentest: Makefile
   	pkgsrc/sysutils/strigi: Makefile
   	pkgsrc/sysutils/virt-manager: Makefile
   	pkgsrc/sysutils/xcdroast: Makefile
   	pkgsrc/sysutils/xfce4-appfinder: Makefile
   	pkgsrc/sysutils/xfce4-battery-plugin: Makefile buildlink3.mk
   	pkgsrc/sysutils/xfce4-cpugraph-plugin: Makefile buildlink3.mk
   	pkgsrc/sysutils/xfce4-diskperf-plugin: Makefile buildlink3.mk
   	pkgsrc/sysutils/xfce4-fsguard-plugin: Makefile buildlink3.mk
   	pkgsrc/sysutils/xfce4-genmon-plugin: Makefile buildlink3.mk
   	pkgsrc/sysutils/xfce4-netload-plugin: Makefile buildlink3.mk
   	pkgsrc/sysutils/xfce4-quicklauncher-plugin: Makefile buildlink3.mk
   	pkgsrc/sysutils/xfce4-systemload-plugin: Makefile buildlink3.mk
   	pkgsrc/sysutils/xfce4-thunar: Makefile buildlink3.mk
   	pkgsrc/sysutils/xfce4-volman: Makefile
   	pkgsrc/sysutils/xfce4-xarchiver: Makefile buildlink3.mk
   	pkgsrc/sysutils/xfce4-xkb-plugin: Makefile buildlink3.mk
   	pkgsrc/textproc/FlightCrew: Makefile
   	pkgsrc/textproc/OdfConverter: Makefile
   	pkgsrc/textproc/dikt: Makefile
   	pkgsrc/textproc/ebview: Makefile
   	pkgsrc/textproc/gnome-spell: Makefile buildlink3.mk
   	pkgsrc/textproc/gnome-subtitles: Makefile
   	pkgsrc/textproc/gtkspell: Makefile buildlink3.mk
   	pkgsrc/textproc/py-gdick: Makefile
   	pkgsrc/textproc/soprano: Makefile buildlink3.mk
   	pkgsrc/textproc/xfce4-dict-plugin: Makefile buildlink3.mk
   	pkgsrc/time/cairo-clock: Makefile
   	pkgsrc/time/evolution-webcal: Makefile
   	pkgsrc/time/gchore: Makefile
   	pkgsrc/time/gdeskcal: Makefile
   	pkgsrc/time/gnotime: Makefile
   	pkgsrc/time/gtodo: Makefile
   	pkgsrc/time/gtodo-applet: Makefile
   	pkgsrc/time/hamster-applet: Makefile
   	pkgsrc/time/ktimer: Makefile
   	pkgsrc/time/planner: Makefile
   	pkgsrc/time/rsibreak: Makefile
   	pkgsrc/time/xfce4-datetime-plugin: Makefile buildlink3.mk
   	pkgsrc/time/xfce4-orage: Makefile
   	pkgsrc/time/xfce4-timer-plugin: Makefile buildlink3.mk
   	pkgsrc/wm/afterstep: Makefile
   	pkgsrc/wm/awesome: Makefile
   	pkgsrc/wm/bmpanel2: Makefile
   	pkgsrc/wm/compiz: Makefile buildlink3.mk
   	pkgsrc/wm/compiz-fusion-plugins-extra: Makefile
   	pkgsrc/wm/compiz-fusion-plugins-main: Makefile buildlink3.mk
   	pkgsrc/wm/e16menuedit2: Makefile
   	pkgsrc/wm/fluxconf: Makefile
   	pkgsrc/wm/metacity: Makefile buildlink3.mk
   	pkgsrc/wm/metisse: Makefile
   	pkgsrc/wm/openbox: Makefile
   	pkgsrc/wm/oroborox: Makefile
   	pkgsrc/wm/sawfish: Makefile
   	pkgsrc/wm/selectwm: Makefile
   	pkgsrc/wm/wbar: Makefile
   	pkgsrc/wm/wmakerconf: Makefile
   	pkgsrc/wm/xfce4-wm: Makefile
   	pkgsrc/wm/xfce4-wm-themes: Makefile
   	pkgsrc/www/amaya: Makefile
   	pkgsrc/www/ap22-dnssd: Makefile
   	pkgsrc/www/bluefish: Makefile
   	pkgsrc/www/browser-bookmarks-menu: Makefile
   	pkgsrc/www/drivel: Makefile
   	pkgsrc/www/epiphany: Makefile buildlink3.mk
   	pkgsrc/www/epiphany-extensions: Makefile
   	pkgsrc/www/firefox: Makefile
   	pkgsrc/www/firefox10: Makefile
   	pkgsrc/www/firefox36: Makefile
   	pkgsrc/www/gtkhtml314: Makefile
   	pkgsrc/www/icedtea-web: Makefile
   	pkgsrc/www/kazehakase: Makefile
   	pkgsrc/www/kdewebdev4: Makefile
   	pkgsrc/www/libgtkhtml: Makefile buildlink3.mk
   	pkgsrc/www/libmediawiki: Makefile buildlink3.mk
   	pkgsrc/www/liferea: Makefile
   	pkgsrc/www/liferea-current: Makefile
   	pkgsrc/www/midori: Makefile
   	pkgsrc/www/netsurf: Makefile
   	pkgsrc/www/nspluginwrapper: Makefile
   	pkgsrc/www/seamonkey: Makefile
   	pkgsrc/www/webkit-gtk: Makefile buildlink3.mk
   	pkgsrc/www/webkit-gtk3: Makefile buildlink3.mk
   	pkgsrc/x11/alacarte: Makefile
   	pkgsrc/x11/antiright: Makefile
   	pkgsrc/x11/avant-window-navigator: Makefile
   	pkgsrc/x11/clisp-gtk2: Makefile
   	pkgsrc/x11/deskmenu: Makefile
   	pkgsrc/x11/devilspie: Makefile
   	pkgsrc/x11/eekboard: Makefile buildlink3.mk
   	pkgsrc/x11/fast-user-switch-applet: Makefile
   	pkgsrc/x11/fltk13: Makefile
   	pkgsrc/x11/ftmenu: Makefile
   	pkgsrc/x11/gdm: Makefile
   	pkgsrc/x11/gnome-applets: Makefile
   	pkgsrc/x11/gnome-control-center: Makefile buildlink3.mk
   	pkgsrc/x11/gnome-desktop: Makefile buildlink3.mk
   	pkgsrc/x11/gnome-desktop-sharp: Makefile buildlink3.mk
   	pkgsrc/x11/gnome-desktop3: Makefile buildlink3.mk
   	pkgsrc/x11/gnome-mag: Makefile buildlink3.mk
   	pkgsrc/x11/gnome-panel: Makefile buildlink3.mk
   	pkgsrc/x11/gnome-screensaver: Makefile
   	pkgsrc/x11/gnome-session: Makefile
   	pkgsrc/x11/gnome-sharp: Makefile buildlink3.mk
   	pkgsrc/x11/gnome-terminal: Makefile
   	pkgsrc/x11/gnome-themes: Makefile
   	pkgsrc/x11/gnome-themes-extras: Makefile
   	pkgsrc/x11/gnopernicus: Makefile
   	pkgsrc/x11/grandr_applet: Makefile
   	pkgsrc/x11/gromit: Makefile
   	pkgsrc/x11/gselt: Makefile
   	pkgsrc/x11/gtk-sharp: Makefile
   	pkgsrc/x11/gtk2: Makefile buildlink3.mk
   	pkgsrc/x11/gtk2+extra: Makefile buildlink3.mk
   	pkgsrc/x11/gtk2-chtheme: Makefile
   	pkgsrc/x11/gtk2-engines: Makefile buildlink3.mk
   	pkgsrc/x11/gtk2-engines-bluecurve: Makefile
   	pkgsrc/x11/gtk2-engines-murrine: Makefile
   	pkgsrc/x11/gtk2-theme-switch: Makefile
   	pkgsrc/x11/gtk3: Makefile buildlink3.mk
   	pkgsrc/x11/gtkada: Makefile buildlink3.mk
   	pkgsrc/x11/gtkglarea2: Makefile buildlink3.mk
   	pkgsrc/x11/gtkmm: Makefile buildlink3.mk
   	pkgsrc/x11/gtkmm-utils: Makefile buildlink3.mk
   	pkgsrc/x11/gtksourceview: Makefile buildlink3.mk
   	pkgsrc/x11/gtksourceview-sharp2: Makefile
   	pkgsrc/x11/gtksourceview2: Makefile buildlink3.mk
   	pkgsrc/x11/gtksourceview3: Makefile buildlink3.mk
   	pkgsrc/x11/gtkterm2: Makefile
   	pkgsrc/x11/hot-babe: Makefile
   	pkgsrc/x11/kactivities: Makefile buildlink3.mk
   	pkgsrc/x11/kde-baseapps4: Makefile
   	pkgsrc/x11/kde-runtime4: Makefile buildlink3.mk
   	pkgsrc/x11/kde-workspace4: Makefile buildlink3.mk
   	pkgsrc/x11/kdelibs4: Makefile buildlink3.mk
   	pkgsrc/x11/konsole: Makefile
   	pkgsrc/x11/lablgtk: Makefile buildlink3.mk
   	pkgsrc/x11/libdesktop-agnostic: Makefile buildlink3.mk
   	pkgsrc/x11/libgnomekbd: Makefile
   	pkgsrc/x11/libunique: Makefile buildlink3.mk
   	pkgsrc/x11/libunique3: Makefile
   	pkgsrc/x11/libxfce4gui: Makefile buildlink3.mk
   	pkgsrc/x11/libxfce4menu: Makefile buildlink3.mk
   	pkgsrc/x11/libxfce4util: Makefile
   	pkgsrc/x11/matchbox-panel-manager: Makefile
   	pkgsrc/x11/mlterm: Makefile
   	pkgsrc/x11/nucleo: Makefile buildlink3.mk
   	pkgsrc/x11/p5-Alien-wxWidgets: Makefile
   	pkgsrc/x11/p5-Wx: Makefile buildlink3.mk
   	pkgsrc/x11/p5-Wx-Perl-ProcessStream: Makefile
   	pkgsrc/x11/p5-gtk2: Makefile buildlink3.mk
   	pkgsrc/x11/py-gnome2: Makefile buildlink3.mk
   	pkgsrc/x11/py-gnome2-desktop: Makefile buildlink3.mk
   	pkgsrc/x11/py-gnome2-extras: Makefile buildlink3.mk
   	pkgsrc/x11/py-gtk2: Makefile buildlink3.mk
   	pkgsrc/x11/py-gtksourceview: Makefile
   	pkgsrc/x11/py-kde4: Makefile
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   	pkgsrc/x11/py-qt4-qscintilla: Makefile buildlink3.mk
   	pkgsrc/x11/py-vte: Makefile buildlink3.mk
   	pkgsrc/x11/py-wxWidgets: Makefile buildlink3.mk
   	pkgsrc/x11/qt4-creator: Makefile
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   	pkgsrc/x11/qt4-mng: Makefile
   	pkgsrc/x11/qt4-mysql: Makefile
   	pkgsrc/x11/qt4-pgsql: Makefile
   	pkgsrc/x11/qt4-qdbus: Makefile buildlink3.mk
   	pkgsrc/x11/qt4-qscintilla: Makefile buildlink3.mk
   	pkgsrc/x11/qt4-sqlite3: Makefile
   	pkgsrc/x11/qt4-tiff: Makefile
   	pkgsrc/x11/qt4-tools: Makefile buildlink3.mk
   	pkgsrc/x11/qwt-qt4: Makefile buildlink3.mk
   	pkgsrc/x11/qwt6-qt4: Makefile buildlink3.mk
   	pkgsrc/x11/qwtplot3d-qt4: Makefile buildlink3.mk
   	pkgsrc/x11/rep-gtk2: Makefile
   	pkgsrc/x11/rox-session: Makefile
   	pkgsrc/x11/ruby-gnome2-gtk: Makefile buildlink3.mk
   	pkgsrc/x11/ruby-gnome2-gtksourceview2: Makefile
   	pkgsrc/x11/ruby-gnome2-vte: Makefile
   	pkgsrc/x11/ruby-wxruby: Makefile
   	pkgsrc/x11/sakura: Makefile
   	pkgsrc/x11/tint2: Makefile
   	pkgsrc/x11/vte: Makefile buildlink3.mk
   	pkgsrc/x11/vte029: Makefile buildlink3.mk
   	pkgsrc/x11/wxGTK24: Makefile buildlink3.mk
   	pkgsrc/x11/wxGTK26: Makefile buildlink3.mk
   	pkgsrc/x11/wxGTK26-contrib: Makefile buildlink3.mk
   	pkgsrc/x11/wxGTK28: Makefile buildlink3.mk
   	pkgsrc/x11/wxGTK28-contrib: Makefile buildlink3.mk
   	pkgsrc/x11/wxhaskell: Makefile
   	pkgsrc/x11/x11vnc: Makefile
   	pkgsrc/x11/xdialog: Makefile
   	pkgsrc/x11/xfce4-clipman-plugin: Makefile buildlink3.mk
   	pkgsrc/x11/xfce4-desktop: Makefile
   	pkgsrc/x11/xfce4-exo: Makefile buildlink3.mk
   	pkgsrc/x11/xfce4-eyes-plugin: Makefile buildlink3.mk
   	pkgsrc/x11/xfce4-gtk2-engine: Makefile
   	pkgsrc/x11/xfce4-notes-plugin: Makefile buildlink3.mk
   	pkgsrc/x11/xfce4-panel: Makefile buildlink3.mk
   	pkgsrc/x11/xfce4-places-plugin: Makefile buildlink3.mk
   	pkgsrc/x11/xfce4-screenshooter-plugin: Makefile buildlink3.mk
   	pkgsrc/x11/xfce4-session: Makefile buildlink3.mk
   	pkgsrc/x11/xfce4-settings: Makefile
   	pkgsrc/x11/xfce4-terminal: Makefile
   	pkgsrc/x11/xfce4-utils: Makefile
   	pkgsrc/x11/xlockmore: Makefile
   	pkgsrc/x11/xpad: Makefile
   	pkgsrc/x11/xplanet: Makefile
   	pkgsrc/x11/xscreensaver-demo: Makefile
   	pkgsrc/x11/xvidcap-gtk2: Makefile
   	pkgsrc/x11/zenity: Makefile

   Log Message:
   Mass recursive bump after the dependence fix of the "cairo" package
   requested by Thomas Klausner.


   To generate a diff of this commit:
   cvs rdiff -u -r1.8 -r1.9 pkgsrc/archivers/ark/Makefile
   cvs rdiff -u -r1.120 -r1.121 pkgsrc/archivers/file-roller/Makefile
   cvs rdiff -u -r1.21 -r1.22 pkgsrc/audio/albumplayer/Makefile
   cvs rdiff -u -r1.114 -r1.115 pkgsrc/audio/amarok/Makefile
   cvs rdiff -u -r1.27 -r1.28 pkgsrc/audio/amarok-kde3/Makefile
   cvs rdiff -u -r1.22 -r1.23 pkgsrc/audio/ario/Makefile
   cvs rdiff -u -r1.27 -r1.28 pkgsrc/audio/audacious/Makefile
   cvs rdiff -u -r1.19 -r1.20 pkgsrc/audio/audacious/buildlink3.mk
   cvs rdiff -u -r1.33 -r1.34 pkgsrc/audio/audacious-plugins/Makefile
   cvs rdiff -u -r1.77 -r1.78 pkgsrc/audio/audacity/Makefile
   cvs rdiff -u -r1.17 -r1.18 pkgsrc/audio/aumix-gtk/Makefile
   cvs rdiff -u -r1.48 -r1.49 pkgsrc/audio/bmp/Makefile
   cvs rdiff -u -r1.22 -r1.23 pkgsrc/audio/bmp/buildlink3.mk
   cvs rdiff -u -r1.29 -r1.30 pkgsrc/audio/bmp-esound/Makefile
   cvs rdiff -u -r1.12 -r1.13 pkgsrc/audio/bmp-flac/Makefile
   cvs rdiff -u -r1.18 -r1.19 pkgsrc/audio/bmp-mac/Makefile
   cvs rdiff -u -r1.26 -r1.27 pkgsrc/audio/bmp-musepack/Makefile
   cvs rdiff -u -r1.16 -r1.17 pkgsrc/audio/bmp-pulse/Makefile
   cvs rdiff -u -r1.14 -r1.15 pkgsrc/audio/bmp-scrobbler/Makefile
   cvs rdiff -u -r1.47 -r1.48 pkgsrc/audio/bmpx/Makefile
   cvs rdiff -u -r1.16 -r1.17 pkgsrc/audio/bsl/Makefile
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   cvs rdiff -u -r1.75 -r1.76 pkgsrc/audio/easytag/Makefile
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   cvs rdiff -u -r1.25 -r1.26 pkgsrc/x11/qt4-mysql/Makefile
   cvs rdiff -u -r1.33 -r1.34 pkgsrc/x11/qt4-pgsql/Makefile
   cvs rdiff -u -r1.27 -r1.28 pkgsrc/x11/qt4-qdbus/Makefile
   cvs rdiff -u -r1.16 -r1.17 pkgsrc/x11/qt4-qdbus/buildlink3.mk
   cvs rdiff -u -r1.20 -r1.21 pkgsrc/x11/qt4-qscintilla/Makefile
   cvs rdiff -u -r1.14 -r1.15 pkgsrc/x11/qt4-qscintilla/buildlink3.mk
   cvs rdiff -u -r1.28 -r1.29 pkgsrc/x11/qt4-sqlite3/Makefile
   cvs rdiff -u -r1.25 -r1.26 pkgsrc/x11/qt4-tiff/Makefile
   cvs rdiff -u -r1.53 -r1.54 pkgsrc/x11/qt4-tools/Makefile
   cvs rdiff -u -r1.23 -r1.24 pkgsrc/x11/qt4-tools/buildlink3.mk
   cvs rdiff -u -r1.19 -r1.20 pkgsrc/x11/qwt-qt4/Makefile
   cvs rdiff -u -r1.16 -r1.17 pkgsrc/x11/qwt-qt4/buildlink3.mk
   cvs rdiff -u -r1.4 -r1.5 pkgsrc/x11/qwt6-qt4/Makefile
   cvs rdiff -u -r1.3 -r1.4 pkgsrc/x11/qwt6-qt4/buildlink3.mk
   cvs rdiff -u -r1.14 -r1.15 pkgsrc/x11/qwtplot3d-qt4/Makefile \
       pkgsrc/x11/qwtplot3d-qt4/buildlink3.mk
   cvs rdiff -u -r1.21 -r1.22 pkgsrc/x11/rep-gtk2/Makefile
   cvs rdiff -u -r1.30 -r1.31 pkgsrc/x11/rox-session/Makefile
   cvs rdiff -u -r1.35 -r1.36 pkgsrc/x11/ruby-gnome2-gtk/Makefile
   cvs rdiff -u -r1.20 -r1.21 pkgsrc/x11/ruby-gnome2-gtk/buildlink3.mk
   cvs rdiff -u -r1.24 -r1.25 pkgsrc/x11/ruby-gnome2-gtksourceview2/Makefile
   cvs rdiff -u -r1.27 -r1.28 pkgsrc/x11/ruby-gnome2-vte/Makefile
   cvs rdiff -u -r1.7 -r1.8 pkgsrc/x11/ruby-wxruby/Makefile
   cvs rdiff -u -r1.9 -r1.10 pkgsrc/x11/sakura/Makefile
   cvs rdiff -u -r1.18 -r1.19 pkgsrc/x11/tint2/Makefile
   cvs rdiff -u -r1.91 -r1.92 pkgsrc/x11/vte/Makefile
   cvs rdiff -u -r1.32 -r1.33 pkgsrc/x11/vte/buildlink3.mk
   cvs rdiff -u -r1.3 -r1.4 pkgsrc/x11/vte029/Makefile \
       pkgsrc/x11/vte029/buildlink3.mk
   cvs rdiff -u -r1.21 -r1.22 pkgsrc/x11/wxGTK24/Makefile
   cvs rdiff -u -r1.24 -r1.25 pkgsrc/x11/wxGTK24/buildlink3.mk
   cvs rdiff -u -r1.17 -r1.18 pkgsrc/x11/wxGTK26/Makefile
   cvs rdiff -u -r1.16 -r1.17 pkgsrc/x11/wxGTK26/buildlink3.mk
   cvs rdiff -u -r1.14 -r1.15 pkgsrc/x11/wxGTK26-contrib/Makefile
   cvs rdiff -u -r1.17 -r1.18 pkgsrc/x11/wxGTK26-contrib/buildlink3.mk
   cvs rdiff -u -r1.25 -r1.26 pkgsrc/x11/wxGTK28/Makefile
   cvs rdiff -u -r1.18 -r1.19 pkgsrc/x11/wxGTK28/buildlink3.mk
   cvs rdiff -u -r1.17 -r1.18 pkgsrc/x11/wxGTK28-contrib/Makefile \
       pkgsrc/x11/wxGTK28-contrib/buildlink3.mk
   cvs rdiff -u -r1.37 -r1.38 pkgsrc/x11/wxhaskell/Makefile
   cvs rdiff -u -r1.42 -r1.43 pkgsrc/x11/x11vnc/Makefile
   cvs rdiff -u -r1.33 -r1.34 pkgsrc/x11/xdialog/Makefile
   cvs rdiff -u -r1.26 -r1.27 pkgsrc/x11/xfce4-clipman-plugin/Makefile
   cvs rdiff -u -r1.30 -r1.31 pkgsrc/x11/xfce4-clipman-plugin/buildlink3.mk
   cvs rdiff -u -r1.47 -r1.48 pkgsrc/x11/xfce4-desktop/Makefile
   cvs rdiff -u -r1.22 -r1.23 pkgsrc/x11/xfce4-exo/Makefile
   cvs rdiff -u -r1.17 -r1.18 pkgsrc/x11/xfce4-exo/buildlink3.mk
   cvs rdiff -u -r1.14 -r1.15 pkgsrc/x11/xfce4-eyes-plugin/Makefile
   cvs rdiff -u -r1.16 -r1.17 pkgsrc/x11/xfce4-eyes-plugin/buildlink3.mk
   cvs rdiff -u -r1.23 -r1.24 pkgsrc/x11/xfce4-gtk2-engine/Makefile
   cvs rdiff -u -r1.28 -r1.29 pkgsrc/x11/xfce4-notes-plugin/Makefile
   cvs rdiff -u -r1.32 -r1.33 pkgsrc/x11/xfce4-notes-plugin/buildlink3.mk
   cvs rdiff -u -r1.50 -r1.51 pkgsrc/x11/xfce4-panel/Makefile
   cvs rdiff -u -r1.37 -r1.38 pkgsrc/x11/xfce4-panel/buildlink3.mk
   cvs rdiff -u -r1.17 -r1.18 pkgsrc/x11/xfce4-places-plugin/Makefile
   cvs rdiff -u -r1.19 -r1.20 pkgsrc/x11/xfce4-places-plugin/buildlink3.mk
   cvs rdiff -u -r1.14 -r1.15 pkgsrc/x11/xfce4-screenshooter-plugin/Makefile
   cvs rdiff -u -r1.16 -r1.17 \
       pkgsrc/x11/xfce4-screenshooter-plugin/buildlink3.mk
   cvs rdiff -u -r1.38 -r1.39 pkgsrc/x11/xfce4-session/Makefile
   cvs rdiff -u -r1.33 -r1.34 pkgsrc/x11/xfce4-session/buildlink3.mk
   cvs rdiff -u -r1.17 -r1.18 pkgsrc/x11/xfce4-settings/Makefile
   cvs rdiff -u -r1.24 -r1.25 pkgsrc/x11/xfce4-terminal/Makefile
   cvs rdiff -u -r1.45 -r1.46 pkgsrc/x11/xfce4-utils/Makefile
   cvs rdiff -u -r1.97 -r1.98 pkgsrc/x11/xlockmore/Makefile
   cvs rdiff -u -r1.40 -r1.41 pkgsrc/x11/xpad/Makefile
   cvs rdiff -u -r1.58 -r1.59 pkgsrc/x11/xplanet/Makefile
   cvs rdiff -u -r1.46 -r1.47 pkgsrc/x11/xscreensaver-demo/Makefile
   cvs rdiff -u -r1.22 -r1.23 pkgsrc/x11/xvidcap-gtk2/Makefile
   cvs rdiff -u -r1.70 -r1.71 pkgsrc/x11/zenity/Makefile

-------------------------------------------------------------------
   Module Name:	pkgsrc
   Committed By:	tron
   Date:		Tue Oct  2 17:18:28 UTC 2012

   Modified Files:
   	pkgsrc/graphics/cairo: Makefile buildlink3.mk

   Log Message:
   Revert unintentional revision bump.


   To generate a diff of this commit:
   cvs rdiff -u -r1.106 -r1.107 pkgsrc/graphics/cairo/Makefile
   cvs rdiff -u -r1.45 -r1.46 pkgsrc/graphics/cairo/buildlink3.mk
@
text
@d1 1
a1 1
# $NetBSD$
d10 1
a10 1
PKGREVISION=	2
@


1.46
log
@Update to Asterisk 1.8.16.0.  This release is to fix AST-2012-012 and
AST-2012-013, and some general bugs.


The Asterisk Development Team has announced the release of Asterisk 1.8.16.0.

The release of Asterisk 1.8.16.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* --- AST-2012-012: Resolve AMI User Unauthorized Shell Access through
      ExternalIVR

* --- AST-2012-013: Resolve ACL rules being ignored during calls by
      some IAX2 peers

* --- Handle extremely out of order RFC 2833 DTMF

* --- Resolve severe memory leak in CEL logging modules.

* --- Only re-create an SRTP session when needed; respond with correct
      crypto policy

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.16.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.45 2012/09/09 06:04:01 jnemeth Exp $
d10 1
@


1.45
log
@Update to Asterisk 1.8.15.1.  This fixes AST-2012-012 and AST-2012-013.

The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are
released as versions 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones.

The release of Asterisk 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones
resolve the following two issues:

* A permission escalation vulnerability in Asterisk Manager Interface.  This
  would potentially allow remote authenticated users the ability to execute
  commands on the system shell with the privileges of the user running the
  Asterisk application.  Please note that the README-SERIOUSLY.bestpractices.txt
  file delivered with Asterisk has been updated due to this and other related
  vulnerabilities fixed in previous versions of Asterisk.

* When an IAX2 call is made using the credentials of a peer defined in a
  dynamic Asterisk Realtime Architecture (ARA) backend, the ACL rules for that
  peer are not applied to the call attempt. This allows for a remote attacker
  who is aware of a peer's credentials to bypass the ACL rules set for that
  peer.

These issues and their resolutions are described in the security advisories.

For more information about the details of these vulnerabilities, please read
security advisories AST-2012-012 and AST-2012-013, which were released at the
same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.15.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2012-012.pdf
 * http://downloads.asterisk.org/pub/security/AST-2012-013.pdf

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.44 2012/09/07 19:16:18 adam Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.15.1
@


1.44
log
@Revbump after updating graphics/cairo
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.43 2012/08/19 18:41:10 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.15.0
a9 1
PKGREVISION=	1
@


1.43
log
@Fix the build issue where some people were seeing chan_mgcp.so
being built and others weren't by detecting the situation when it
would be built and adjusting the PLIST accordingly.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.42 2012/08/03 08:22:53 jnemeth Exp $
d10 1
@


1.42
log
@Update to Asterisk 1.8.15.0; this is a general bug fix release:

The release of Asterisk 1.8.15.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* --- Fix deadlock potential with ast_set_hangupsource() calls.

* --- Fix request routing issue when outboundproxy is used.

* --- Make the address family filter specific to the transport.

* --- Fix NULL pointer segfault in ast_sockaddr_parse()

* --- Do not perform install on existing directories

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.15.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.41 2012/07/17 00:29:33 jnemeth Exp $
d104 8
@


1.41
log
@- revert previous
  - this package is marked OWNER= for a reason!
  - need to figure out why chan_mgcp is built only in some situations
    instead of adding gross hacks

- upgrade to Asterisk 1.8.14.1: this is a bugfix release

The release of Asterisk 1.8.14.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

* --- Remove a superfluous and dangerous freeing of an SSL_CTX.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.14.1

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.40 2012/07/15 09:00:00 marino Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.14.1
@


1.40
log
@comms/asterisk18: PLIST+= chan_mgcp.so (DragonFly Only)

Defined new PLIST.mgcp variable for new file:
lib/asterisks/modules/chan_mgcp.so
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.39 2012/07/15 07:49:27 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.14.0
a103 5
PLIST_VARS+=	mgcp
.if ${OPSYS} == "DragonFly"
PLIST.mgcp=	yes
.endif

@


1.39
log
@Update to Asterisk 1.8.14.0:  this is a bugfix release

The Asterisk Development Team has announced the release of Asterisk 1.8.14.0.

The release of Asterisk 1.8.14.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* --- format_mp3: Fix a possible crash in mp3_read().

* --- Fix local channel chains optimizing themselves out of a call.

* --- Update a peer's LastMsgsSent when the peer is notified of
      waiting messages

* --- Prevent sip_pvt refleak when an ast_channel outlasts its
      corresponding sip_pvt.

* --- Send more accurate identification information in dialog-info SIP
      NOTIFYs.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.14.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.38 2012/07/06 19:51:58 jnemeth Exp $
d104 5
@


1.38
log
@Upgrade to Asterisk 1.8.13.1 -- this fixes two security issues:
AST-2012-010 and AST-2012-011

The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are
released as versions 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones.

The release of Asterisk 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones
resolve the following two issues:

* If Asterisk sends a re-invite and an endpoint responds to the re-invite with
  a provisional response but never sends a final response, then the SIP dialog
  structure is never freed and the RTP ports for the call are never released. If
  an attacker has the ability to place a call, they could create a denial of
  service by using all available RTP ports.

* If a single voicemail account is manipulated by two parties simultaneously,
  a condition can occur where memory is freed twice causing a crash.

These issues and their resolution are described in the security advisories.

For more information about the details of these vulnerabilities, please read
security advisories AST-2012-010 and AST-2012-011, which were released at the
same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.13.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2012-010.pdf
 * http://downloads.asterisk.org/pub/security/pST-2012-011.pdf

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.37 2012/06/14 07:45:20 sbd Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.13.1
@


1.37
log
@Recursive PKGREVISION bump for libxml2 buildlink addition.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.36 2012/06/05 00:15:34 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.13.0
a9 1
PKGREVISION=	1
@


1.36
log
@Update to 1.8.13.0:

The Asterisk Development Team has announced the release of Asterisk
1.8.13.0.

The release of Asterisk 1.8.13.0 resolves several issues reported
by the community and would have not been possible without your
participation.  Thank you!

The following is a sample of the issues resolved in this release:

* --- Turn off warning message when bind address is set to any.

* --- Prevent overflow in calculation in ast_tvdiff_ms on 32-bit
      machines

* --- Make DAHDISendCallreroutingFacility wait 5 seconds for a reply
      before disconnecting the call.

* --- Fix recalled party B feature flags for a failed DTMF atxfer.

* --- Fix DTMF atxfer running h exten after the wrong bridge ends.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.13.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.35 2012/06/03 22:58:50 jnemeth Exp $
d10 1
@


1.35
log
@Update to Asterisk 1.8.12.2:  this update fixes fixes AST-2012-007
and AST-2012-008 along with some general bug fixes.

----- 1.8.12.1 -----

The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.11 and Asterisk 1.8 and 10. The available
security releases are released as versions 1.8.11-cert2, 1.8.12.1,
and 10.4.1.

The release of Asterisk 1.8.11-cert2, 1.8.12.1, and 10.4.1 resolve
the following two issues:

* A remotely exploitable crash vulnerability exists in the IAX2
  channel driver if an established call is placed on hold without
  a suggested music class. Asterisk will attempt to use an invalid
  pointer to the music on hold class name, potentially causing a
  crash.

* A remotely exploitable crash vulnerability was found in the Skinny
  (SCCP) Channel driver. When an SCCP client closes its connection
  to the server, a pointer in a structure is set to NULL.  If the
  client was not in the on-hook state at the time the connection
  was closed, this pointer is later dereferenced. This allows remote
  authenticated connections the ability to cause a crash in the
  server, denying services to legitimate users.

These issues and their resolution are described in the security
advisories.

For more information about the details of these vulnerabilities,
please read security advisories AST-2012-007 and AST-2012-008,
which were released at the same time as this announcement.

For a full list of changes in the current releases, please see the
ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.12.1
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.4.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2012-007.pdf
 * http://downloads.asterisk.org/pub/security/AST-2012-008.pdf

Thank you for your continued support of Asterisk!

----- 1.8.12.2 -----

The Asterisk Development Team has announced the release of Asterisk
1.8.12.2.

The release of Asterisk 1.8.12.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

* --- Resolve crash in subscribing for MWI notifications
  (Closes issue ASTERISK-19827. Reported by B. R)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.12.2

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.34 2012/05/06 05:40:50 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.12.2
d178 2
a179 4
	res_stun_monitor.conf rpt.conf rtp.conf say.conf sip.conf	\
	sip_notify.conf skinny.conf sla.conf smdi.conf telcordia-1.adsi	\
	udptl.conf unistim.conf usbradio.conf users.conf voicemail.conf	\
	vpb.conf
@


1.34
log
@Update to Asterisk 1.8.12.0: this is a bugfix release

pkgsrc changes:

- set OPTIMIZE to -O3 as levels above are poorly defined and can
  cause problems
- maintain current patch namimg convention

-----

The Asterisk Development Team has announced the release of Asterisk 1.8.12.0.

The release of Asterisk 1.8.12.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

* --- Prevent chanspy from binding to zombie channels

* --- Fix Dial m and r options and forked calls generating warnings
      for voice frames.

* --- Remove ISDN hold restriction for non-bridged calls.

* --- Fix copying of CDR(accountcode) to local channels.

* --- Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors

* --- Eliminate double close of file descriptor in manager.c

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.12.0

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.33 2012/05/04 16:06:14 joerg Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.12.0
@


1.33
log
@Don't override optimizer settings with absurd levels.
Fix inline definitions to work with C99 compiler.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.32 2012/04/30 02:33:21 jnemeth Exp $
d6 1
a6 1
DISTNAME=	asterisk-1.8.11.1
d93 1
a93 1
MAKE_FLAGS+=		OPTIMIZE=
@


1.32
log
@Update to Asterisk 1.8.11.1.  This fixes AST-2012-004, AST-2012-005,
and AST-2012-006.

The Asterisk Development Team has announced security releases for
Asterisk 1.6.2 , 1.8, and 10. The available security releases are
released as versions 1.6.2.24, 1.8.11.1, and 10.3.1.

The release of Asterisk 1.6.2.24, 1.8.11.1, and 10.3.1 resolve the
following two issues:

 * A permission escalation vulnerability in Asterisk Manager
   Interface.  This would potentially allow remote authenticated
   users the ability to execute commands on the system shell with
   the privileges of the user running the Asterisk application.

 * A heap overflow vulnerability in the Skinny Channel driver.
   The keypad button message event failed to check the length of
   a fixed length buffer before appending a received digit to the
   end of that buffer.  A remote authenticated user could send
   sufficient keypad button message events that th e buffer would
   be overrun.

In addition, the release of Asterisk 1.8.11.1 and 10.3.1 resolve
the following issue:

 * A remote crash vulnerability in the SIP channel driver when
   processing UPDATE requests.  If a SIP UPDATE request was received
   indicating a connected line update after a channel was terminated
   but before the final destruction of the associated SIP dialog,
   Asterisk would attempt a connected line update on a non-existing
   channel, causing a crash.

These issues and their resolution are described in the security
advisories.

For more information about the details of these vulnerabilities,
please read security advisories AST-2012-004, AST-2012-005, and
AST-2012-006, which were released at the same time as this
announcement.

For a full list of changes in the current releases, please see the
ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.24
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.11.1
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.3.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2012-004.pdf
 * http://downloads.asterisk.org/pub/security/AST-2012-005.pdf
 * http://downloads.asterisk.org/pub/security/AST-2012-006.pdf

Thank you for your continued support of Asterisk!
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.31 2012/04/27 12:31:38 obache Exp $
d93 1
@


1.31
log
@Recursive bump from icu shlib major bumped to 49.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.30 2012/04/18 01:33:24 manu Exp $
d6 1
a6 2
DISTNAME=	asterisk-1.8.11.0
PKGREVISION=	2
@


1.30
log
@Link chan_oss.so with libossaudio to avoid startup load failure because
Undefined PLT symbol "_oss_ioctl" (symnum = 64)
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.29 2012/04/07 20:10:45 jnemeth Exp $
d7 1
a7 1
PKGREVISION=	1
@


1.29
log
@Update to Asterisk 1.8.11.0:

pkgsrc change: eliminate ilbc option now that the iLBC codec is always built

The Asterisk Development Team has announced the release of Asterisk 1.8.11.0.

The release of Asterisk 1.8.11.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

* --- Fix potential buffer overrun and memory leak when executing "sip
      show peers"

* --- Fix ACK routing for non-2xx responses.

* --- Remove possible segfaults from res_odbc by adding locks around
      usage of odbc handle

* --- Fix blind transfer parking issues if the dialed extension is not
      recognized as a parking extension.

* --- Copy CDR variables when set during a bridge

* --- push 'outgoing' flag from sig_XXX up to chan_dahdi

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.11.0

Thank you for your continued support of Asterisk!
@
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@


1.28
log
@Update to 1.8.10.1:  this fixes AST-2012-002 and AST-2012-003.

pkgsrc changes: adapt to having iLBC coded included in the asterisk
tarball and newer version of sounds tarball.

----- 1.8.10.0 -----

The Asterisk Development Team has announced the release of Asterisk 1.8.10.0.

The release of Asterisk 1.8.10.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* --- Prevent outbound SIP NOTIFY packets from displaying a port of 0 ---

* --- Include iLBC source code for distribution with Asterisk ---

* --- Fix callerid of originated calls ---

* --- Fix outbound DTMF for inband mode of chan_ooh323 ---

* --- Create and initialize udptl only when dialog requests image media ---

* --- Don't prematurely stop SIP session timer ---

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.10.0

Thank you for your continued support of Asterisk!

----- 1.8.10.1 -----

The Asterisk Development Team has announced security releases for
Asterisk 1.4, 1.6.2, 1.8, and 10. The available security releases
are released as versions 1.4.44, 1.6.2.23, 1.8.10.1, and 10.2.1.

The release of Asterisk 1.8.10.1 and 10.2.1 resolve two issues.
First, they resolve the issue in app_milliwatt, wherein a buffer
can potentially be overrun on the stack, but no remote code execution
is possible.  Second, they resolve an issue in HTTP AMI where digest
authentication information can be used to overrun a buffer on the
stack, allowing for code injection and execution.

These issues and their resolution are described in the security
advisory.

For more information about the details of these vulnerabilities,
please read the security advisories AST-2012-002 and AST-2012-003,
which were released at the same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLogs:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.10.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2012-002.pdf
 * http://downloads.asterisk.org/pub/security/AST-2012-003.pdf

Thank you for your continued support of Asterisk!
@
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DISTNAME=	asterisk-1.8.10.1
@


1.28.2.1
log
@Pullup ticket #3715 - requested by dholland
comms/asterisk10 and comms/asterisk18 update/build fixes.

Revisions pulled up:
- comms/asterisk10/Makefile                                     1.11
- comms/asterisk10/PLIST                                        1.5
- comms/asterisk10/distinfo                                     1.8
- comms/asterisk10/options.mk                                   1.4
- comms/asterisk10/patches/patch-configure                      1.3
- comms/asterisk10/patches/patch-main_features.c                1.2
- comms/asterisk18/Makefile                                     1.29
- comms/asterisk18/PLIST                                        1.15
- comms/asterisk18/distinfo                                     1.24
- comms/asterisk18/options.mk                                   1.10
- comms/asterisk18/patches/patch-aq                             1.9
- comms/asterisk18/patches/patch-ay                             1.4

---
   Module Name:	pkgsrc
   Committed By:	jnemeth
   Date:		Sat Apr  7 20:05:57 UTC 2012

   Modified Files:
   	pkgsrc/comms/asterisk10: Makefile PLIST distinfo options.mk
   	pkgsrc/comms/asterisk10/patches: patch-configure patch-main_features.c

   Log Message:
   Update to Asterisk 10.3.0:

   pkgsrc change: eliminate ilbc option now that iLBC codec is always built

   The Asterisk Development Team has announced the release of Asterisk 10.3.0.

   The release of Asterisk 10.3.0 resolves several issues reported by the
   community and would have not been possible without your participation.
   Thank you!

   The following are the issues resolved in this release:

   * --- Fix potential buffer overrun and memory leak when executing "sip
         show peers"

   * --- Fix ACK routing for non-2xx responses.

   * --- Remove possible segfaults from res_odbc by adding locks around
         usage of odbc handle

   * --- Fix blind transfer parking issues if the dialed extension is not
         recognized as a parking extension.

   * --- Copy CDR variables when set during a bridge

   * --- push 'outgoing' flag from sig_XXX up to chan_dahdi

   For a full list of changes in this release, please see the ChangeLog:

   http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.3.0

   Thank you for your continued support of Asterisk!

---
   Module Name:	pkgsrc
   Committed By:	jnemeth
   Date:		Sat Apr  7 20:10:45 UTC 2012

   Modified Files:
   	pkgsrc/comms/asterisk18: Makefile PLIST distinfo options.mk
   	pkgsrc/comms/asterisk18/patches: patch-aq patch-ay

   Log Message:
   Update to Asterisk 1.8.11.0:

   pkgsrc change: eliminate ilbc option now that the iLBC codec is always built

   The Asterisk Development Team has announced the release of Asterisk 1.8.11.0.

   The release of Asterisk 1.8.11.0 resolves several issues reported by the
   community and would have not been possible without your participation.
   Thank you!

   The following are the issues resolved in this release:

   * --- Fix potential buffer overrun and memory leak when executing "sip
         show peers"

   * --- Fix ACK routing for non-2xx responses.

   * --- Remove possible segfaults from res_odbc by adding locks around
         usage of odbc handle

   * --- Fix blind transfer parking issues if the dialed extension is not
         recognized as a parking extension.

   * --- Copy CDR variables when set during a bridge

   * --- push 'outgoing' flag from sig_XXX up to chan_dahdi

   For a full list of changes in this release, please see the ChangeLog:

   http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.11.0

   Thank you for your continued support of Asterisk!
@
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DISTNAME=	asterisk-1.8.11.0
@


1.27
log
@More pcre PKGREVISION bumps.
@
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DISTNAME=	asterisk-1.8.9.3
a9 1
PKGREVISION=	1
d192 1
a192 1
	${TAR} xzf ${WRKSRC}/sounds/asterisk-core-sounds-en-gsm-1.4.21.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
@


1.26
log
@Update to 1.8.9.3:

pkgsrc changes:

- maintain patch naming convention
- detect kqueue properly

The Asterisk Development Team has announced the release of Asterisk 1.8.9.3.

The release of Asterisk 1.8.9.3 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

* --- Fix ACK routing for non-2xx responses.
  (Closes issue ASTERISK-19389. Reported by: Karsten Wemheuer)

* --- Fix regressions with regards to route-set creation on early dialogs ---
  (Closes issue ASTERISK-19358. Reported-by: Karsten Wemheuer)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.3

Thank you for your continued support of Asterisk!
@
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@


1.25
log
@Fix build on SunOS.
@
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DISTNAME=	asterisk-1.8.9.2
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.if ${OPSYS} != "SunOS"
@


1.24
log
@Update to Asterisk 1.8.9.2:

The release of Asterisk 1.8.9.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolve
@
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@


1.23
log
@Update to 1.8.9.1:

The release of Asterisk 1.8.9.1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* --- Fixes deadlocks occuring in chan_agent ---

* --- Ensure entering T.38 passthrough does not cause an infinite loop ---

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.1

Thank you for your continued support of Asterisk!
@
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DISTNAME=	asterisk-1.8.9.1
@


1.22
log
@Revbump for
a) tiff update to 4.0 (shlib major change)
b) glib2 update 2.30.2 (adds libffi dependency to buildlink3.mk)

Enjoy.
@
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a9 1
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@


1.21
log
@Update to Asterisk 1.8.9.0:

The Asterisk Development Team is pleased to announce the release of
Asterisk 1.8.9.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.9.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* AST-2012-001: prevent crash when an SDP offer
  is received with an encrypted video stream when support for video
  is disabled and res_srtp is loaded.  (closes issue ASTERISK-19202)
  Reported by: Catalin Sanda

* Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop.  Failing
  to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
  causes the loop to exit prematurely. This causes a variety of negative side
  effects, depending on when the loop exits. This patch handles the frame by
  essentially swallowing the frame in the local loop, as the current channel
  drivers expect the RTP bridge to handle the frame, and, in the case of the
  local bridge loop, no additional action is necessary.
  (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
  by: Matt Jordan

* Fix timing source dependency issues with MOH.  Prior to this patch,
  res_musiconhold existed at the same module priority level as the timing
  sources that it depends on.  This would cause a problem when music on
  hold was reloaded, as the timing source could be changed after
  res_musiconhold was processed. This patch adds a new module priority
  level, AST_MODPRI_TIMING, that the various timing modules are now loaded
  at. This now occurs before loading other resource modules, such
  that the timing source is guaranteed to be set prior to resolving
  the timing source dependencies.
  (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H,
  Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
  Patched by elguero

* Fix RTP reference leak.  If a blind transfer were initiated using a
  REFER without a prior reINVITE to place the call on hold, AND if Asterisk
  were sending RTCP reports, then there was a reference leak for the
  RTP instance of the transferrer.
  (closes issue ASTERISK-19192) Reported by: Tyuta Vitali

* Fix blind transfers from failing if an 'h' extension
  is present.  This prevents the 'h' extension from being run on the
  transferee channel when it is transferred via a native transfer
  mechanism such as SIP REFER.  (closes issue ASTERISK-19173) Reported
  by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by
  Mark Michelson (license 5049)

* Restore call progress code for analog ports. Extracting sig_analog
  from chan_dahdi lost call progress detection functionality.  Fix
  analog ports from considering a call answered immediately after
  dialing has completed if the callprogress option is enabled.
  (closes issue ASTERISK-18841)
  Reported by: Richard Miller Patched by Richard Miller

* Fix regression that 'rtp/rtcp set debup ip' only works when a port
  was also specified.
  (closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by:
  Walter Doekes

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.0

Thank you for your continued support of Asterisk!
@
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1.20
log
@Update to Asterisk 1.8.8.2.  This fixes AST-2010-001:

               Asterisk Project Security Advisory - AST-2012-001

   +------------------------------------------------------------------------+
   |       Product        | Asterisk                                        |
   |----------------------+-------------------------------------------------|
   |       Summary        | SRTP Video Remote Crash Vulnerability           |
   |----------------------+-------------------------------------------------|
   |  Nature of Advisory  | Denial of Service                               |
   |----------------------+-------------------------------------------------|
   |    Susceptibility    | Remote unauthenticated sessions                 |
   |----------------------+-------------------------------------------------|
   |       Severity       | Moderate                                        |
   |----------------------+-------------------------------------------------|
   |    Exploits Known    | No                                              |
   |----------------------+-------------------------------------------------|
   |     Reported On      | 2012-01-15                                      |
   |----------------------+-------------------------------------------------|
   |     Reported By      | Catalin Sanda                                   |
   |----------------------+-------------------------------------------------|
   |      Posted On       | 2012-01-19                                      |
   |----------------------+-------------------------------------------------|
   |   Last Updated On    | January 19, 2012                                |
   |----------------------+-------------------------------------------------|
   |   Advisory Contact   | Joshua Colp < jcolp AT digium DOT com >         |
   |----------------------+-------------------------------------------------|
   |       CVE Name       |                                                 |
   +------------------------------------------------------------------------+

   +------------------------------------------------------------------------+
   | Description | An attacker attempting to negotiate a secure video       |
   |             | stream can crash Asterisk if video support has not been  |
   |             | enabled and the res_srtp Asterisk module is loaded.      |
   +------------------------------------------------------------------------+

   +------------------------------------------------------------------------+
   | Resolution | Upgrade to one of the versions of Asterisk listed in the  |
   |            | "Corrected In" section, or apply a patch specified in the |
   |            | "Patches" section.                                        |
   +------------------------------------------------------------------------+

   +------------------------------------------------------------------------+
   |                           Affected Versions                            |
   |------------------------------------------------------------------------|
   |            Product            | Release Series |                       |
   |-------------------------------+----------------+-----------------------|
   |     Asterisk Open Source      |     1.8.x      | All versions          |
   |-------------------------------+----------------+-----------------------|
   |     Asterisk Open Source      |      10.x      | All versions          |
   +------------------------------------------------------------------------+

   +------------------------------------------------------------------------+
   |                              Corrected In                              |
   |------------------------------------------------------------------------|
   |                 Product                  |           Release           |
   |------------------------------------------+-----------------------------|
   |           Asterisk Open Source           |           1.8.8.2           |
   |------------------------------------------+-----------------------------|
   |           Asterisk Open Source           |           10.0.1            |
   +------------------------------------------------------------------------+

   +------------------------------------------------------------------------+
   |                                Patches                                 |
   |------------------------------------------------------------------------|
   |                             SVN URL                             |Branch|
   |-----------------------------------------------------------------+------|
   |http://downloads.asterisk.org/pub/security/AST-2012-001-1.8.diff |v1.8  |
   |-----------------------------------------------------------------+------|
   |http://downloads.asterisk.org/pub/security/AST-2012-001-10.diff  |v10   |
   +------------------------------------------------------------------------+

   +------------------------------------------------------------------------+
   |   Links   | https://issues.asterisk.org/jira/browse/ASTERISK-19202     |
   +------------------------------------------------------------------------+

   +------------------------------------------------------------------------+
   | Asterisk Project Security Advisories are posted at                     |
   | http://www.asterisk.org/security                                       |
   |                                                                        |
   | This document may be superseded by later versions; if so, the latest   |
   | version will be posted at                                              |
   | http://downloads.digium.com/pub/security/AST-2012-001.pdf and          |
   | http://downloads.digium.com/pub/security/AST-2012-001.html             |
   +------------------------------------------------------------------------+

   +------------------------------------------------------------------------+
   |                            Revision History                            |
   |------------------------------------------------------------------------|
   |      Date       |       Editor       |         Revisions Made          |
   |-----------------+--------------------+---------------------------------|
   | 12-01-19        | Joshua Colp        | Initial release                 |
   +------------------------------------------------------------------------+

               Asterisk Project Security Advisory - AST-2012-001
              Copyright (c) 2012 Digium, Inc. All Rights Reserved.
  Permission is hereby granted to distribute and publish this advisory in its
                           original, unaltered form.
@
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@


1.19
log
@PR/35369 -- David Wetzel -- add support for speex codec (enabled by default)
@
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DISTNAME=	asterisk-1.8.8.1
PKGREVISION=	1
@


1.18
log
@Update to Asterisk 1.8.8.1.

share/doc/asterisk/AST.{txt,pdf} has been replaced with
share/doc/asterisk/Asterisk_Admin_Guide.  You will need a browser
to read the latter.

----- Asterisk 1.8.8.1 -----

The release of Asterisk 1.8.8.1 resolves a regression introduced
in Asterisk 1.8.8.0 reported by the community, and would have not
been possible without your participation.  Thank you!

The following is the issue resolved in this release:

* Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop

  Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local
  bridge loop causes the loop to exit prematurely.  This causes a
  variety of negative side effects, which may include having Music
  On Hold failing during a SIP Hold.

For a full description of the changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.1

Thank you for your continued support of Asterisk!

----- Asterisk 1.8.8.0 -----

The release of Asterisk 1.8.8.0 resolves several issues reported
by the community and would have not been possible without your
participation.  Thank you!

The following is a sample of the issues resolved in this release:

* Updated SIP 484 handling; added Incomplete control frame
   When a SIP phone uses the dial application and receives a 484
   Address Incomplete response, if overlapped dialing is enabled
   for SIP, then the 484 Address Incomplete is forwarded back to
   the SIP phone and the HANGUPCAUSE channel variable is set to
   28. Previously, the Incomplete application dialplan logic was
   automatically triggered; now, explicit dialplan usage of the
   application is required.

* Prevent IAX2 from getting IPv6 addresses via DNS
   IAX2 does not support IPv6 and getting such addresses from DNS
   can cause error messages on the remote end involving bad IPv4
   address casts in the presence of IPv6/IPv4 tunnels.

* Fix bad RTP media bridges in directmedia calls on peers separated by
  multiple Asterisk nodes.

* Fix crashes in ast_rtcp_write()

* Fix for incorrect voicemail duration in external notifications.
   This patch fixes an issue where the voicemail duration was being
   reported with a duration significantly less than the actual
   sound file duration.

* Prevent segfault if call arrives before Asterisk is fully booted.

* Fix remote Crash Vulnerability in SIP channel driver (AST-2011-012)
     http://downloads.asterisk.org/pub/security/AST-2011-012.pdf

* Fix locking order in app_queue.c which caused deadlocks

* Fix regression in configure script for libpri capability checks

* Prevent BLF subscriptions from causing deadlocks.

* Fix deadlock if peer is destroyed while sending MWI notice.

* Fix issue with setting defaultenabled on categories that are already
  enabled by default.

* Don't crash on INFO automon request with no channel
     AST-2011-014. When automon was enabled in features.conf, it
     was possible to crash Asterisk by sending an INFO request if
     no channel had been created yet.

* Fixed crash from orphaned MWI subscriptions in chan_sip
   This patch resolves the issue where MWI subscriptions are orphaned
   by subsequent SIP SUBSCRIBE messages.

* Default to nat=yes; warn when nat in general and peer differ
     AST-2011-013. It is possible to enumerate SIP usernames when
     the general and user/peer nat settings differ in whether to
     respond to the port a request is sent from or the port listed
     for responses in the Via header.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0

Thank you for your continued support of Asterisk!
@
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1.17
log
@This update is to fix AST-2011-013 and AST-2011-014.

               Asterisk Project Security Advisory - AST-2011-013

         Product        Asterisk
         Summary        Possible remote enumeration of SIP endpoints with
                        differing NAT settings
    Nature of Advisory  Unauthorized data disclosure
      Susceptibility    Remote unauthenticated sessions
         Severity       Minor
      Exploits Known    Yes
       Reported On      2011-07-18
       Reported By      Ben Williams
        Posted On
     Last Updated On    December 7, 2011
     Advisory Contact   Terry Wilson <twilson at digium.com>

         CVE Name

    Description  It is possible to enumerate SIP usernames when the general
                 and user/peer NAT settings differ in whether to respond to
                 the port a request is sent from or the port listed for
                 responses in the Via header. In 1.4 and 1.6.2, this would
                 mean if one setting was nat=yes or nat=route and the other
                 was either nat=no or nat=never. In 1.8 and 10, this would
                 mean when one was nat=force_rport or nat=yes and the other
                 was nat=no or nat=comedia.

    Resolution  Handling NAT for SIP over UDP requires the differing
                behavior introduced by these options.

                To lessen the frequency of unintended username disclosure,
                the default NAT setting was changed to always respond to the
                port from which we received the request-the most commonly
                used option.

                Warnings were added on startup to inform administrators of
                the risks of having a SIP peer configured with a different
                setting than that of the general setting. The documentation
                now strongly suggests that peers are no longer configured
                for NAT individually, but through the global setting in the
                "general" context.

                               Affected Versions
                Product              Release Series
         Asterisk Open Source             All        All versions

                                  Corrected In
     As this is more of an issue with SIP over UDP in general, there is no
     fix supplied other than documentation on how to avoid the problem. The
        default NAT setting has been changed to what we believe the most
      commonly used setting for the respective version in Asterisk 1.4.43,
                             1.6.2.21, and 1.8.7.2.

            Links

    Asterisk Project Security Advisories are posted at
    http://www.asterisk.org/security

    This document may be superseded by later versions; if so, the latest
    version will be posted at
    http://downloads.digium.com/pub/security/AST-2011-013.pdf and
    http://downloads.digium.com/pub/security/AST-2011-013.html

                                Revision History
           Date                 Editor                 Revisions Made

               Asterisk Project Security Advisory - AST-2011-013
              Copyright (c) 2011 Digium, Inc. All Rights Reserved.
  Permission is hereby granted to distribute and publish this advisory in its
                           original, unaltered form.

     __________________________________________________________________

               Asterisk Project Security Advisory - AST-2011-014

         Product        Asterisk
         Summary        Remote crash possibility with SIP and the "automon"
                        feature enabled
    Nature of Advisory  Remote crash vulnerability in a feature that is
                        disabled by default
      Susceptibility    Remote unauthenticated sessions
         Severity       Moderate
      Exploits Known    Yes
       Reported On      November 2, 2011
       Reported By      Kristijan Vrban
        Posted On       2011-11-03
     Last Updated On    December 7, 2011
     Advisory Contact   Terry Wilson <twilson at digium.com>

         CVE Name

    Description  When the "automon" feature is enabled in features.conf, it
                 is possible to send a sequence of SIP requests that cause
                 Asterisk to dereference a NULL pointer and crash.

    Resolution  Applying the referenced patches that check that the pointer
                is not NULL before accessing it will resolve the issue. The
                "automon" feature can be disabled in features.conf as a
                workaround.

                               Affected Versions
                Product              Release Series
         Asterisk Open Source           1.6.2.x      All versions
         Asterisk Open Source            1.8.x       All versions

                                  Corrected In
                   Product                              Release
            Asterisk Open Source                   1.6.2.21, 1.8.7.2

                                     Patches
                              Download URL                            Revision
   http://downloads.asterisk.org/pub/security/AST-2011-014-1.6.2.diff 1.6.2.20
   http://downloads.asterisk.org/pub/security/AST-2011-014-1.8.diff   1.8.7.1

            Links

    Asterisk Project Security Advisories are posted at
    http://www.asterisk.org/security

    This document may be superseded by later versions; if so, the latest
    version will be posted at
    http://downloads.digium.com/pub/security/AST-2011-014.pdf and
    http://downloads.digium.com/pub/security/AST-2011-014.html

                                Revision History
           Date                 Editor                 Revisions Made

               Asterisk Project Security Advisory - AST-2011-014
              Copyright (c) 2011 Digium, Inc. All Rights Reserved.
  Permission is hereby granted to distribute and publish this advisory in its
                           original, unaltered form.
@
text
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DISTNAME=	asterisk-1.8.7.2
a198 2
	${INSTALL_DATA} ${WRKSRC}/doc/AST.pdf ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/doc/AST.txt ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
d203 1
@


1.16
log
@Recursive bump for graphics/freetype2 buildlink addition.
@
text
@d1 1
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d6 1
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DISTNAME=	asterisk-1.8.7.1
a9 1
PKGREVISION=	1
@


1.15
log
@Update to 1.8.7.1 -- this update fixes AST-2011-012

pkgsrc change:  now what sqlite3 has been imported into NetBSD, enable it

               Asterisk Project Security Advisory - AST-2011-012

          Product         Asterisk
          Summary         Remote crash vulnerability in SIP channel driver
     Nature of Advisory   Remote crash
       Susceptibility     Remote authenticated sessions
          Severity        Critical
       Exploits Known     No
        Reported On       October 4, 2011
        Reported By       Ehsan Foroughi
         Posted On        October 17, 2011
      Last Updated On     October 17, 2011
      Advisory Contact    Terry Wilson <twilson@@digium.com>
          CVE Name        CVE-2011-4063

    Description  A remote authenticated user can cause a crash with a
                 malformed request due to an unitialized variable.

    Resolution  Ensure variables are initialized in all cases when parsing
                the request.

                               Affected Versions
           Product         Release Series
    Asterisk Open Source       1.8.x       All versions
    Asterisk Open Source        10.x       All versions (currently in beta)

                                  Corrected In
                  Product                              Release
            Asterisk Open Source                 1.8.7.1, 10.0.0-rc1

                                    Patches
                             Download URL                           Revision
   http://downloads.asterisk.org/pub/security/AST-2011-012-1.8.diff 1.8
   http://downloads.asterisk.org/pub/security/AST-2011-012-10.diff  10

            Links

    Asterisk Project Security Advisories are posted at
    http://www.asterisk.org/security

    This document may be superseded by later versions; if so, the latest
    version will be posted at
    http://downloads.digium.com/pub/security/AST-2011-012.pdf and
    http://downloads.digium.com/pub/security/AST-2011-012.html

                                Revision History
           Date                 Editor                 Revisions Made

               Asterisk Project Security Advisory - AST-2011-012
              Copyright (c) 2011 Digium, Inc. All Rights Reserved.
  Permission is hereby granted to distribute and publish this advisory in its
                           original, unaltered form.
@
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@


1.14
log
@Update to 1.8.7.0nb1.

This update adds a "jabber" option which is enabled by default.
This option pulls in iksemel which is used by the res_jabber.
Doing this allows chan_jingle (jabber) and chan_gtalk to work.
@
text
@d1 1
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d6 1
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DISTNAME=	asterisk-1.8.7.0
PKGREVISION=	1
d206 1
@


1.13
log
@Update to 1.8.7.0 (mainly bug fixes).

pkgsrc changes:
- adjust for ilbc changes after it was acquired by Google
- install AST.pdf IAX2-security.pdf into share/doc/asterisk

1.8.7.0:
========

The release of Asterisk 1.8.7.0 resolves several issues reported
by the community and would have not been possible without your
participation.  Thank you!

Please note that a significant numbers of changes and fixes have
gone into features.c in this release (call parking, built-in
transfers, call pickup, etc.).

NOTE:

Recently, we were notified that the mechanism included in our
Asterisk source code releases to download and build support for
the iLBC codec had stopped working correctly; a little investigation
revealed that this occurred because of some changes on the
ilbcfreeware.org website. These changes occurred as a result of
Google's acquisition of GIPS, who produced (and provided licenses
for) the iLBC codec.

If you are a user of Asterisk and iLBC together, and you've already
executed a license agreement with GIPS, we believe you can continue
using iLBC with Asterisk. If you are a user of Asterisk and iLBC
together, but you had not executed a license agreement with GIPS,
we encourage you to research the situation and consult with your
own legal representatives to determine what actions you may want
to take (or avoid taking).

More information is available on the Asterisk blog:

http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/

The following is a sample of the issues resolved in this release:

* Added the 'storesipcause' option to sip.conf to allow the user to
   disable the setting of HASH(SIP_CAUSE,) on the channel. Having
   chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant
   performance penalty because of the usage of the MASTER_CHANNEL()
   dialplan function.

   We've decided to disable this feature by default in future 1.8
   versions. This would be an unexpected behavior change for anyone
   depending on that SIP_CAUSE update in their dialplan. Please
   refer to the asterisk-dev mailing list more information:

   http://lists.digium.com/pipermail/asterisk-dev/2011-August/050626.html

* Significant fixes and improvements to parking lots.
   (Closes issues ASTERISK-17183, ASTERISK-17870, ASTERISK-17430,
   ASTERISK-17452, ASTERISK-17452, ASTERISK-15792.)

* Numerous issues have been reported for deadlocks that are caused
   by a blocking read in res_timing_timerfd on a file descriptor
   that will never be written to.

   A change to Asterisk adds some checks to make sure that the
   timerfd is both valid and armed before calling read(). Should
   fix: ASTERISK-18142, ASTERISK-18197, ASTERISK-18166 and possibly
   others.  (In essence, this change should make res_timing_timerfd
   usable.)

* Resolve segfault when publishing device states via XMPP and not connected.
   (Closes issue ASTERISK-18078.)

* Refresh peer address if DNS unavailable at peer creation.
   (Closes issue ASTERISK-18000)

* Fix the missing DAHDI channels when using the newer chan_dahdi.conf
   sections for channel configuration.
   (Closes issue ASTERISK-18496.)

* Remove unnecessary libpri dependency checks in the configure script.
   (Closes issue ASTERISK-18535.)

* Update get_ilbc_source.sh script to work again.
   (Closes issue ASTERISK-18412)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.7.0

Thank you for your continued support of Asterisk!


1.8.6.0:
========

The release of Asterisk 1.8.6.0 resolves several issues reported
by the community and would have not been possible without your
participation.  Thank you!

The following is a sample of the issues resolved in this release:

* Fix an issue with Music on Hold classes losing files in playlist
   when realtime is used.  (Closes issue ASTERISK-17875.)

* Resolve a potential crash in chan_sip when utilizing auth= and
   performing a 'sip reload' from the console.  (Closes issue
   ASTERISK-17939.)

* Address some improper sql statements in res_odbc that would cause
   an update to fail on realtime peers due to trying to set as
   "(NULL)" rather than an actual NULL.  (Closes issue ASTERISK-17791.)

* Resolve issue where 403 Forbidden would always be sent maximum
   number of times regardless to receipt of ACK.

* Resolve issue where if a call to MeetMe includes both the dynamic(D)
   and always request PIN(P) options, MeetMe will ask for the PIN
   two times:  once for creating the conference and once for entering
   the conference.

* Fix New Zealand indications profile based on
   http://www.telepermit.co.nz/TNA102.pdf
   (Closes issue ASTERISK-16263.)

* Segfault in shell_helper in func_shell.c
   (Closes issue ASTERISK-18109.)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.6.0

Thank you for your continued support of Asterisk!
@
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@


1.12
log
@Bump PKGREVISION for perl update.
@
text
@d1 1
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# $NetBSD: Makefile,v 1.11 2011/07/16 21:35:11 jnemeth Exp $
d6 1
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DISTNAME=	asterisk-1.8.5.0
PKGREVISION=	1
d199 1
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@


1.11
log
@Update to Asterisk 1.8.5.0: this is a general bug fix release

The release of Asterisk 1.8.5.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* Fix Deadlock with attended transfer of SIP call

* Fixes thread blocking issue in the sip TCP/TLS implementation.

* Be more tolerant of what URI we accept for call completion PUBLISH requests.

* Fix a nasty chanspy bug which was causing a channel leak every time a spied on
  channel made a call.

* This patch fixes a bug with MeetMe behavior where the 'P' option for always
  prompting for a pin is ignored for the first caller.

* Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If
  the call that the dialplan started an AGI script for is hungup while the AGI
  script is in the middle of a command then the AGI script is not notified of
  the hangup.

* Resolve issue where leaving a voicemail, the MWI message is never sent. The
  same thing happens when checking a voicemail and marking it as read.

* Resolve issue where wait for leader with Music On Hold allows crosstalk
  between participants. Parenthesis in the wrong position. Regression from issue
  #14365 when expanding conference flags to use 64 bits.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5.0

Thank you for your continued support of Asterisk!
@
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@


1.10
log
@Update to Asterisk 1.8.4.4 (fixes AST-2011-011):

               Asterisk Project Security Advisory - AST-2011-011

   +------------------------------------------------------------------------+
   |      Product       | Asterisk                                          |
   |--------------------+---------------------------------------------------|
   |      Summary       | Possible enumeration of SIP users due to          |
   |                    | differing authentication responses                |
   |--------------------+---------------------------------------------------|
   | Nature of Advisory | Unauthorized data disclosure                      |
   |--------------------+---------------------------------------------------|
   |   Susceptibility   | Remote unauthenticated sessions                   |
   |--------------------+---------------------------------------------------|
   |      Severity      | Moderate                                          |
   |--------------------+---------------------------------------------------|
   |   Exploits Known   | No                                                |
   |--------------------+---------------------------------------------------|
   |      CVE Name      | CVE-2011-2536                                     |
   +------------------------------------------------------------------------+

   +------------------------------------------------------------------------+
   | Description | Asterisk may respond differently to SIP requests from an |
   |             | invalid SIP user than it does to a user configured on    |
   |             | the system, even when the alwaysauthreject option is set |
   |             | in the configuration. This can leak information about    |
   |             | what SIP users are valid on the Asterisk system.         |
   +------------------------------------------------------------------------+

   +------------------------------------------------------------------------+
   | Resolution | Respond to SIP requests from invalid and valid SIP users  |
   |            | in the same way. Asterisk 1.4 and 1.6.2 do not respond    |
   |            | identically by default due to backward-compatibility      |
   |            | reasons, and must have alwaysauthreject=yes set in        |
   |            | sip.conf. Asterisk 1.8 defaults to alwaysauthreject=yes.  |
   |            |                                                           |
   |            | IT IS ABSOLUTELY IMPERATIVE that users of Asterisk 1.4    |
   |            | and 1.6.2 set alwaysauthreject=yes in the general section |
   |            | of sip.conf.                                              |
   +------------------------------------------------------------------------+
@
text
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d126 1
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	${TAR} xzf ${WRKSRC}/sounds/asterisk-core-sounds-en-gsm-1.4.20.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
@


1.9
log
@Upgrade to 1.8.4.2.  This fixes several security issues including:
AST-2011-002, AST-2011-003, AST-2011-004, AST-2011-005, AST-2011-006,
and AST-2011-007.

pkgsrc changes:
- add patch for autosupport script; == -> =
- patch configure to not unconditionally set PBX_LAUNCHD=1
  - this allows res_timing_kqueue.so to build

This last change brings a timing source to NetBSD which allows IAX
trunking and allows the bridging modules to work, a rather major
piece that was missing.  Note that I haven't extensively tested
it.  But, have at it...

===========================================================================
1.8.4.2:

The Asterisk Development Team has announced the release of Asterisk
version 1.8.4.2, which is a security release for Asterisk 1.8.

The release of Asterisk 1.8.4.2 resolves an issue with SIP URI parsing
which can lead to a remotely exploitable crash:

     Remote Crash Vulnerability in SIP channel driver (AST-2011-007)

The issue and resolution is described in the AST-2011-007 security
advisory.

For more information about the details of this vulnerability, please
read the security advisory AST-2011-007, which was released at the same
time as this announcement.

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.2

Security advisory AST-2011-007 is available at:

http://downloads.asterisk.org/pub/security/AST-2011-007.pdf

===========================================================================
1.8.4.1:

The Asterisk Development Team has announced the release of Asterisk 1.8.4.1.

The release of Asterisk 1.8.4.1 resolves several issues reported by the
community. Without your help this release would not have been possible.
Thank you!

Below is a list of issues resolved in this release:

 * Fix our compliance with RFC 3261 section 18.2.2. (aka Cisco phone fix)

 * Resolve a change in IPv6 header parsing due to the Cisco phone fix issue.
   This issue was found and reported by the Asterisk test suite.

 * Resolve potential crash when using SIP TLS support.

 * Improve reliability when using SIP TLS.

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4.1

===========================================================================
1.8.4:

The Asterisk Development Team has announced the release of Asterisk 1.8.4.

The release of Asterisk 1.8.4 resolves several issues reported by the community.
Without your help this release would not have been possible. Thank you!

Below is a sample of the issues resolved in this release:

 * Use SSLv23_client_method instead of old SSLv2 only.

 * Resolve crash in ast_mutex_init()

 * Resolution of several DTMF based attended transfer issues.

   NOTE: Be sure to read the ChangeLog for more information about these changes.

 * Resolve deadlocks related to device states in chan_sip

 * Resolve an issue with the Asterisk manager interface leaking memory when
   disabled.

 * Support greetingsfolder as documented in voicemail.conf.sample.

 * Fix channel redirect out of MeetMe() and other issues with channel softhangup

 * Fix voicemail sequencing for file based storage.

 * Set hangup cause in local_hangup so the proper return code of 486 instead of
   503 when using Local channels when the far sides returns a busy. Also affects
   CCSS in Asterisk 1.8+.

 * Fix issues with verbose messages not being output to the console.

 * Fix Deadlock with attended transfer of SIP call

Includes changes per AST-2011-005 and AST-2011-006
For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4

Information about the security releases are available at:

http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

===========================================================================
1.8.3.3:

The Asterisk Development Team has announced security releases for Asterisk
branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3.

The releases of Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 resolve two
issues:

* File Descriptor Resource Exhaustion (AST-2011-005)
* Asterisk Manager User Shell Access (AST-2011-006)

The issues and resolutions are described in the AST-2011-005 and AST-2011-006
security advisories.

For more information about the details of these vulnerabilities, please read the
security advisories AST-2011-005 and AST-2011-006, which were released at the
same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.3

Security advisory AST-2011-005 and AST-2011-006 are available at:

http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

===========================================================================
1.8.3.2:

he Asterisk Development Team has announced security releases for Asterisk
branches 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.6.1.24, 1.6.2.17.2, and 1.8.3.2.

** This is a re-release of Asterisk 1.6.1.23, 1.6.2.17.1 and 1.8.3.1 which
    contained a bug which caused duplicate manager entries (issue #18987).

The releases of Asterisk 1.6.1.24, 1.6.2.17.2, and 1.8.3.2 resolve two issues:

  * Resource exhaustion in Asterisk Manager Interface (AST-2011-003)
  * Remote crash vulnerability in TCP/TLS server (AST-2011-004)

The issues and resolutions are described in the AST-2011-003 and AST-2011-004
security advisories.

For more information about the details of these vulnerabilities, please read the
security advisories AST-2011-003 and AST-2011-004, which were released at the
same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.2

Security advisory AST-2011-003 and AST-2011-004 are available at:

http://downloads.asterisk.org/pub/security/AST-2011-003.pdf
http://downloads.asterisk.org/pub/security/AST-2011-004.pdf

===========================================================================
1.8.3.1:

The Asterisk Development Team has announced security releases for Asterisk
branches 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.6.1.23, 1.6.2.17.1, and 1.8.3.1.

The releases of Asterisk 1.6.1.23, 1.6.2.17.1, and 1.8.3.1 resolve two issues:

  * Resource exhaustion in Asterisk Manager Interface (AST-2011-003)
  * Remote crash vulnerability in TCP/TLS server (AST-2011-004)

The issues and resolutions are described in the AST-2011-003 and AST-2011-004
security advisories.

For more information about the details of these vulnerabilities, please read the
security advisories AST-2011-003 and AST-2011-004, which were released at the
same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.1

Security advisory AST-2011-003 and AST-2011-004 are available at:

http://downloads.asterisk.org/pub/security/AST-2011-003.pdf
http://downloads.asterisk.org/pub/security/AST-2011-004.pdf

===========================================================================
1.8.3:

The Asterisk Development Team has announced the release of Asterisk 1.8.3.

The release of Asterisk 1.8.3 resolves several issues reported by the community
and would have not been possible without your participation. Thank you!

The following is a sample of the issues resolved in this release:

* Resolve duplicated data in the AstDB when using DIALGROUP()

* Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses.

* Reworking parsing of mwi => lines to resolve a segfault. Also add a set of
   unit tests for the function that does the parsing.

* When using cdr_pgsql the billsec field was not populated correctly on
   unanswered calls.

* Resolve memory leak in iCalendar and Exchange calendaring modules.

* This version of Asterisk includes the new Compiler Flags option
   BETTER_BACKTRACES which uses libbfd to search for better symbol information
   within both the Asterisk binary, as well as loaded modules, to assist when
   using inline backtraces to track down problems.

* Resolve issue where no Music On Hold may be triggered when using
   res_timing_dahdi.

* Resolve a memory leak when the Asterisk Manager Interface is disabled.

* Reimplemented fax session reservation to reverse the ABI breakage introduced
   in r297486.

* Fix regression that changed behavior of queues when ringing a queue member.

* Resolve deadlock involving REFER.

Additionally, this release has the changes related to security bulletin
AST-2011-002 which can be found at
http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3

===========================================================================
1.8.2.4:

The Asterisk Development Team has announced security releases for Asterisk
branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4.

The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an
issue that when decoding UDPTL packets, multiple stack and heap based arrays can
be made to overflow by specially crafted packets. Systems configured for
T.38 pass through or termination are vulnerable. The issue and resolution are
described in the AST-2011-002 security advisory.

For more information about the details of this vulnerability, please read the
security advisory AST-2011-002, which was released at the same time as this
announcement.

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.4

Security advisory AST-2011-002 is available at:

http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
@
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@


1.8
log
@Let not to change DIST_SUBDIR after bump PKGREVISION to 2.
PR#44914.
@
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d6 2
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DISTNAME=	asterisk-1.8.2.3
#XXX change back with next revision
#DIST_SUBDIR=	${PKGNAME_NOREV}
DIST_SUBDIR=	${PKGNAME_NOREV}nb1
PKGREVISION=	2
a45 1
# Running as non-root will come later
@


1.7
log
@recursive bump from gettext-lib shlib bump.
@
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@


1.6
log
@Bah!  Upstream changed a couple of text files in the distro tarball
without cranking the version number.
@
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@


1.5
log
@Update to 1.8.2.3 -- bug fix release to fix a FAX issue

pkgsrc:  fix issue with patch for detecting sys/atomic.h

The Asterisk Development Team has announced the release of Asterisk 1.8.2.3.

The release of Asterisk 1.8.2.3 resolves the following issue:

  * Reimplemented fax session reservation to reverse the ABI breakage introduced
    in r297486.
    (Reported by Jeremy Kister on the asterisk-users mailing list. Patched by
    mnicholson)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2.3
@
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@


1.4
log
@Update to 1.8.2.2

This is to fix AST-2011-001: Stack buffer overflow in SIP channel driver

               Asterisk Project Security Advisory - AST-2011-001

         Product        Asterisk
         Summary        Stack buffer overflow in SIP channel driver
    Nature of Advisory  Exploitable Stack Buffer Overflow
      Susceptibility    Remote Authenticated Sessions
         Severity       Moderate
      Exploits Known    No
       Reported On      January 11, 2011
       Reported By      Matthew Nicholson
        Posted On       January 18, 2011
     Last Updated On    January 18, 2011
     Advisory Contact   Matthew Nicholson <mnicholson at digium.com>
         CVE Name

   Description When forming an outgoing SIP request while in pedantic mode, a
               stack buffer can be made to overflow if supplied with
               carefully crafted caller ID information. This vulnerability
               also affects the URIENCODE dialplan function and in some
               versions of asterisk, the AGI dialplan application as well.
               The ast_uri_encode function does not properly respect the size
               of its output buffer and can write past the end of it when
               encoding URIs.

For full details, see:

http://downloads.digium.com/pub/security/AST-2011-001.html
@
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@


1.3
log
@Update to 1.8.2:

The release of Asterisk 1.8.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* 'sip notify clear-mwi' needs terminating CRLF.
   (Closes issue #18275. Reported, patched by klaus3000)

* Patch for deadlock from ordering issue between channel/queue locks in
   app_queue (set_queue_variables).
   (Closes issue #18031. Reported by rain. Patched by bbryant)

* Fix cache of device state changes for multiple servers.
   (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested
   by russellb)

* Resolve issue where channel redirect function (CLI or AMI) hangs up the call
   instead of redirecting the call.
   (Closes issue #18171. Reported by: SantaFox)
   (Closes issue #18185. Reported by: kwemheuer)
   (Closes issue #18211. Reported by: zahir_koradia)
   (Closes issue #18230. Reported by: vmarrone)
   (Closes issue #18299. Reported by: mbrevda)
   (Closes issue #18322. Reported by: nerbos)

* Fix reloading of peer when a user is requested. Prevent peer reloading from
   causing multiple MWI subscriptions to be created when using realtime.
   (Closes issue #18342. Reported, patched by nivek.)

* Fix XMPP PubSub-based distributed device state. Initialize pubsubflags to 0
   so res_jabber doesn't think there is already an XMPP connection sending
   device state. Also clean up CLI commands a bit.
   (Closes issue #18272. Reported by klaus3000. Patched by Marquis42)

* Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of
   setting peer->cdr = NULL, set it to not post.
   (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares)

* Fixes issue with outbound google voice calls not working. Thanks to az1234
   and nevermind_quack for their input in helping debug the issue.
   (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2
@
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@


1.2
log
@Update to 1.8.1.1.  This is a minor bugfix update.

The release of Asterisk 1.8.1.1 resolves two issues reported by the community
since the release of Asterisk 1.8.1.

  * Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of
    setting peer->cdr = NULL, set it to not post.
    (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares)

  * Fixes issue with outbound google voice calls not working. Thanks to az1234
    and nevermind_quack for their input in helping debug the issue.
    (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1.1
@
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@


1.2.2.1
log
@Pullup ticket #3336 - requested by gls
comms/asterisk18: security update

Revisions pulled up:
- comms/asterisk18/Makefile			1.3-1.4
- comms/asterisk18/distinfo			1.5-1.6
- comms/asterisk18/patches/patch-aq		1.2
---
Module Name:	pkgsrc
Committed By:	jnemeth
Date:		Sun Jan 16 17:52:43 UTC 2011

Modified Files:
	pkgsrc/comms/asterisk18: Makefile distinfo
	pkgsrc/comms/asterisk18/patches: patch-aq

Log Message:
Update to 1.8.2:

The release of Asterisk 1.8.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* 'sip notify clear-mwi' needs terminating CRLF.
   (Closes issue #18275. Reported, patched by klaus3000)

* Patch for deadlock from ordering issue between channel/queue locks in
   app_queue (set_queue_variables).
   (Closes issue #18031. Reported by rain. Patched by bbryant)

* Fix cache of device state changes for multiple servers.
   (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched,
tested
   by russellb)

* Resolve issue where channel redirect function (CLI or AMI) hangs up
the call
   instead of redirecting the call.
   (Closes issue #18171. Reported by: SantaFox)
   (Closes issue #18185. Reported by: kwemheuer)
   (Closes issue #18211. Reported by: zahir_koradia)
   (Closes issue #18230. Reported by: vmarrone)
   (Closes issue #18299. Reported by: mbrevda)
   (Closes issue #18322. Reported by: nerbos)

* Fix reloading of peer when a user is requested. Prevent peer reloading
from
   causing multiple MWI subscriptions to be created when using realtime.
   (Closes issue #18342. Reported, patched by nivek.)

* Fix XMPP PubSub-based distributed device state. Initialize pubsubflags
to 0
   so res_jabber doesn't think there is already an XMPP connection sending
   device state. Also clean up CLI commands a bit.
   (Closes issue #18272. Reported by klaus3000. Patched by Marquis42)

* Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of
   setting peer->cdr = NULL, set it to not post.
   (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares)

* Fixes issue with outbound google voice calls not working. Thanks to az1234
   and nevermind_quack for their input in helping debug the issue.
   (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2
---
odule Name:	pkgsrc
Committed By:	jnemeth
Date:		Fri Jan 21 07:00:44 UTC 2011

Modified Files:
	pkgsrc/comms/asterisk18: Makefile distinfo

Log Message:
Update to 1.8.2.2

This is to fix AST-2011-001: Stack buffer overflow in SIP channel driver

               Asterisk Project Security Advisory - AST-2011-001

         Product        Asterisk
         Summary        Stack buffer overflow in SIP channel driver
    Nature of Advisory  Exploitable Stack Buffer Overflow
      Susceptibility    Remote Authenticated Sessions
         Severity       Moderate
      Exploits Known    No
       Reported On      January 11, 2011
       Reported By      Matthew Nicholson
        Posted On       January 18, 2011
     Last Updated On    January 18, 2011
     Advisory Contact   Matthew Nicholson <mnicholson at digium.com>
         CVE Name

   Description When forming an outgoing SIP request while in pedantic
mode, a
               stack buffer can be made to overflow if supplied with
               carefully crafted caller ID information. This vulnerability
               also affects the URIENCODE dialplan function and in some
               versions of asterisk, the AGI dialplan application as well..
               The ast_uri_encode function does not properly respect the
size
               of its output buffer and can write past the end of it when
               encoding URIs.

For full details, see:

http://downloads.digium.com/pub/security/AST-2011-001.html
@
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@


1.1
log
@Initial revision
@
text
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DISTNAME=	asterisk-1.8.1
@


1.1.1.1
log
@     Import Asterisk 1.8.1:


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 1.8 is a long term support version (i.e. it will be
supported for four years with an additional year of security only
fixes).  See:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

     What's new:

Asterisk 1.8 is the next major release series of Asterisk.

The release of Asterisk 1.8.0 would not have been possible without the support
and contributions of the community. Since Asterisk 1.6.2, we've had over 500
reporters, more than 300 testers and greater than 200 developers contributed to
this release.

You can find a summary of the work involved with the 1.8.0 release in the
sumary:

http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt

A short list of available features includes:

     * Secure RTP
     * IPv6 Support in the SIP channel driver
     * Connected Party Identification Support
     * Calendaring Integration
     * A new call logging system, Channel Event Logging (CEL)
     * Distributed Device State using Jabber/XMPP PubSub
     * Call Completion Supplementary Services support
     * Advice of Charge support
     * Much, much more!

A full list of new features can be found in the CHANGES file.

http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup

For a full list of changes in the current release candidate, please see the
ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0

-----

The Asterisk Development Team has announced the release of Asterisk 1.8.1.

The release of Asterisk 1.8.1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

* Fix issue when using directmedia. Asterisk needs to limit the codecs offered
   to just the ones that both sides recognize, otherwise they may end up sending
   audio that the other side doesn't understand.
   (Closes issue #17403. Reported, patched by one47. Tested by one47, falves11)

* Resolve issue where Party A in an analog 3-way call would continue to hear
   ringback after party C answers.
   (Patched by rmudgett)

* Fix playback failure when using IAX with the timerfd module.
   (Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler)

* Fix problem with qualify option packets for realtime peers never stopping.
   The option packets not only never stopped, but if a realtime peer was not in
   the peer list multiple options dialogs could accumulate over time.
   (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
   jpeeler)

* Fix issue where it is possible to crash Asterisk by feeding the curl engine
   invalid data.
   (Closes issue #18161. Reported by wdoekes. Patched by tilghman)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
@
text
@@


1.1.1.2
log
@Import Asterisk 18.x as comms/asterisk18.

This is a long term support version.  It is scheduled to go to
security fixes only on October 20th, 2024, and EOL on October 20th,
2025.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.3.0 to Asterisk 18.4.0 ------------
------------------------------------------------------------------------------

logger
------------------
 * The dateformat option in logger.conf will now control the remote
   console (asterisk -r -T) timestamp format.  Previously, dateformat
   only controlled the formatting of the timestamp going to log
   files and the main console (asterisk -c) but only for non-verbose
   messages.

   Internally, Asterisk does not send the logging timestamp with
   verbose messages to console clients. It's up to the Asterisk
   remote consoles to format verbose messages.  Asterisk remote
   consoles previously did not load dateformat from logger.conf.

   Previously there was a non-configurable and hard-coded "%b %e
   %T" dateformat that would be used no matter what on all verbose
   console messages printed on remote consoles.

   Example:
   logger.conf
    dateformat=%F %T.%3q

   # asterisk -rvvv -T
   [2021-03-19 09:54:19.760-0400]  Loading res_stasis_answer.so.
   [Mar 19 09:55:43]     -- Goto (dialExten,s,1)

   Given the following example configuration in logger.conf, Asterisk
   log files and the console, will log verbose messages using the
   given timestamp.  Now ensuring that all remote console messages
   are logged with the same dateformat as other log streams.

   ---
   [general]
   dateformat=%F %T.%3q

   [logfiles]
   console  => notice,warning,error,verbose
   full     => notice,warning,error,debug,verbose
   ---

   Now we have a globally-defined dateformat that will be used
   consistently across the Asterisk main console, remote consoles,
   and log files.

   Now we have consistent logging:

   # asterisk -rvvv -T
   [2021-03-19 09:54:19.760-0400]  Loading res_stasis_answer.so.
   [2021-03-19 09:55:43.920-0400]     -- Goto (dialExten,s,1)

res_pjsip
------------------
 * PJSIP transports can now be partially reloaded safely. This
   allows the local_net and external_* options to be updated without
   restarting Asterisk.

 * PJSIP endpoints can now be configured to skip authentication
   when handling OPTIONS requests by setting the
   allow_unauthenticated_options configuration property to 'yes.'

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.2.2 to Asterisk 18.3.0 ------------
------------------------------------------------------------------------------

app_mixmonitor
------------------
 * app_mixmonitor now sends manager events MixMonitorStart,
   MixMonitorStop and MixMonitorMute when the channel monitoring
   is started, stopped and muted (or unmuted) respectively.

chan_iax2
------------------
 * You can now specify a default "auth" method in the [general]
   section of iax.conf

chan_pjsip, app_transfer
------------------
 * Added TRANSFERSTATUSPROTOCOL variable.  When transfer is performed,
   transfers can pass a protocol specific error code.  Example, in
   SIP 3xx-6xx represent any SIP specific error received when
   performing a REFER.

func_odbc
------------------
 * Introduce an ARGC variable for func_odbc functions, along with
   a minargs per-function configuration option.

   minargs enables enforcing of minimum count of arguments to pass
   to func_odbc, so if you're unconditionally using ARG1 through
   ARG4 then this should be set to 4.  func_odbc will generate an
   error in this case, so for example

   [FOO]
   minargs = 4

   and ODBC_FOO(a,b,c) in dialplan will now error out instead of
   using a potentially leaked ARG4 from Gosub().

   ARGC is needed if you're using optional argument, to verify
   whether or not an argument has been passed, else it's possible
   to use a leaked ARGn from Gosub (app_stack).  So now you can
   safely do ${IF($[${ARGC}>3]?${ARGV}:default value)} kind of
   thing.

res_srtp
------------------
 * SRTP replay protection has been added to res_srtp and
   a new configuration option "srtpreplayprotection" has been added
   to the rtp.conf config file.  For security reasons, the default
   setting is "yes".  Buggy clients may not handle this correctly
   which could result in no, or one way, audio and Asterisk error
   messages like "replay check failed".

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.1.0 to Asterisk 18.2.0 ------------
------------------------------------------------------------------------------

Core
------------------
 * The location where the media cache stores its temporary files
   is no longer hardcoded to /tmp but can now be configured separately
   via the astcachedir config variable in asterisk.conf. To retain
   backwards compatibility, the default location remains /tmp.

app_voicemail
------------------
 * The VoiceMail application can now be configured to send greetings
   and instructions via early media and only answering the channel
   when it is time for the caller to record their message. This
   behavior can be activated by passing the new 'e' option to
   VoiceMail.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.0.0 to Asterisk 18.1.0 ------------
------------------------------------------------------------------------------

Core
------------------
 * Added debug logging categories that allow a user to output debug
   information based on a specified category. This lets the user
   limit, and filter debug output to data relevant to a particular
   context, or topic. For instance the following categories are
   now available for debug logging purposes:

   dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet

   These debug categories can be enable/disable via an Asterisk
   CLI command:

     core set debug category <category>[:<sublevel>] [category[:<sublevel] ...]
     core set debug category off [<category> [<category>] ...]

   If no sub-level is associated all debug statements for a given
   category are output. If a sub-level is given then only those
   statements assigned a value at or below the associated sub-level
   are output.

app_confbridge
------------------
 * app_confbridge now has the ability to force the estimated bitrate
   on an SFU bridge.  To use it, set a bridge profile's remb_behavior
   to "force" and set remb_estimated_bitrate to a rate in bits per
   second.  The remb_estimated_bitrate parameter is ignored if
   remb_behavior is something other than "force".

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------
------------------------------------------------------------------------------

chan_pjsip
------------------
 * The PJSIP_SEND_SESSION_REFRESH dialplan function now issues a
   warning, and returns unsuccessful if it's used on a channel
   prior to answering.

logger
------------------
 * Added a new log formatter called "plain" that always prints
   file, function and line number if available (even for verbose
   messages) and never prints color control characters.  Most
   suitable for file output but can be used for other channels as
   well.

   You use it in logger.conf like so:
   debug => [plain]debug
   console => [plain]error,warning,debug,notice,pjsip_history
   messages => [plain]warning,error,verbose

------------------------------------------------------------------------------
--- New functionality introduced in Asterisk 18.0.0 --------------------------
------------------------------------------------------------------------------

Core
------------------
 * The Streams API becomes the home for the core ACN capabilities.
   These include...

    * Parsing and formatting of codec negotation preferences.
    * Resolving pending streams and topologies with those configured
      using configured preferences.
    * Utility functions for creating string representations of
      streams, topologies, and negotiation preferences.

   For codec negotiation preferences:
    * Added ast_stream_codec_prefs_parse() which takes a string
      representation of codec negotiation preferences, which may
      come from a pjsip endpoint for example, and populates a
      ast_stream_codec_negotiation_prefs structure.
    * Added ast_stream_codec_prefs_to_str() which does the reverse.
    * Added many functions to parse individual parameter name
      and value strings to their respectrive enum values, and the
      reverse.

   For streams:
    * Added ast_stream_create_resolved() which takes a "live" stream
      and resolves it with a configured stream and the negotiation
      preferences to create a new stream.
    * Added ast_stream_to_str() which create a string representation
      of a stream suitable for debug or display purposes.

   For topology:
    * Added ast_stream_topology_create_resolved() which takes a
      "live" topology and resolves it, stream by stream, with a
      configured topology stream and the negotiation preferences
      to create a new topology.
    * Added ast_stream_topology_to_str() which create a string
      representation of a topology suitable for debug or display
      purposes.
    * Renamed ast_format_caps_from_topology() to
      ast_stream_topology_get_formats() to be more consistent with
      the existing ast_stream_get_formats().

   Additional changes:
    * A new function ast_format_cap_append_names() appends the
      results to the ast_str buffer instead of replacing buffer
      contents.

app_bridgeaddchan
------------------
 * The BridgeAdd application now behaves more like the Bridge
   application.  The application now sets the BRIDGERESULT channel
   variable to indicate what happened when the channel resumes in
   dialplan.  This is instead of hanging up the channel on failure
   conditions.

res_pjsip
------------------
 * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
   have been added to res_pjsip endpoints that specify the preferred
   order of codecs to use between those received/sent in an SDP
   offer and those set in the endpoint configuration.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------
------------------------------------------------------------------------------

AMI
------------------
 * You can now specify an optional 'Content-Type' as an argument
   for the Asterisk SendText manager action.

ARI
------------------
 * A new parameter 'inhibitConnectedLineUpdates' is now available
   in the 'bridges.addChannel' call. This prevents the identity of
   the newly connected channel from being presented to other bridge
   members.

ARI Channels
------------------
 * The Channel resource has a new sub-resource "externalMedia".
   This allows an application to create a channel for the sole
   purpose of exchanging media with an external server.  Once
   created, this channel could be placed into a bridge with existing
   channels to allow the external server to inject audio into the
   bridge or receive audio from the bridge.  See
   https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
   for more information.

Core
------------------
 * H.265/HEVC is now a supported video codec and it can be used by
   specifying "h265" in the allow line.  Please note however, that
   handling of the additional SDP parameters described in RFC 7798
   section 7.2 is not yet supported.

Features
------------------
 * Adds support for AudioSocket, a very simple bidirectional audio
   streaming protocol. There are both channel and application
   interfaces.

   A description of the protocol can be found on the referenced
   wiki page. A short talk about the reasons and implementation
   can be found on YouTube at the link provided.

   ARI support has also been added via the existing "externalMedia"
   ARI functionality. The UUID is specified using the arbitrary
   "data" field.

   Wiki: https://wiki.asterisk.org/wiki/display/AST/AudioSocket
   YouTube: https://www.youtube.com/watch?v=tjduXbZZEgI

Messaging
------------------
 * In order to reduce the amount of AMI and ARI events generated,
   the global "Message/ast_msg_queue" channel can be set to suppress
   it's normal channel housekeeping events such as "Newexten",
   "VarSet", etc. This can greatly reduce load on the manager and
   ARI applications when the Digium Phone Module for Asterisk is
   in use.  To enable, set "hide_messaging_ami_events" in asterisk.conf
   to "yes"  In Asterisk versions <18, the default is "no" preserving
   existing behavior.  Beginning with Asterisk 18, the option will
   default to "yes".

STIR/SHAKEN
------------------
 * STIR/SHAKEN support has been added to Asterisk. Configuration
   is done in stir_shaken.conf. There is a sample configuration
   file to help you get started
   (asterisk/configs/samples/stir_shaken.conf.sample).  Once that's
   set up, you can enable STIR/SHAKEN on any endpoint by setting
   stir_shaken to yes on the endpoint configuration object. This
   will add an Identity header on outgoing INVITEs, and check for
   an Identity header on incoming INVITEs. This option has been
   added to Alembic as well.

   The information received on an incoming INVITE can be checked
   using the STIR_SHAKEN dialplan function. There are two variations:

   STIR_SHAKEN(count)
   STIR_SHAKEN(0, verify_result)

   The first variation will tell you how many STIR/SHAKEN results
   are on the channel. The second fetches information for a specific
   result. The first parameter is the index, followed by what
   information you want to retrieve.  The available options are
   'verify_result', 'identity', and 'attestation'.

app_chanisavail
------------------
 * The ChanIsAvail application now tolerates empty positions in
   the supplied device list.  Dialplan can now be simplified by
   not having to check for empty positions in the device list.

app_confbridge
------------------
 * A new bridge profile option, maximum_sample_rate, has been added
   which sets a maximum sample rate that the bridge will be mixed
   at. This allows the bridge to move below the maximum sample rate
   as needed but caps it at the maximum.

 * A new option, "text_messaging", has been added to the user
   profile which allows control over whether text messaging is
   enabled or disabled for a user. If enabled (the default) text
   messages will be sent to the user. If disabled no text messages
   will be sent to the user.

app_dial
------------------
 * The Dial application now tolerates empty positions in the supplied
   destination list.  Dialplan can now be simplified by not having
   to check for empty positions in the destination list.  If there
   are no endpoints to dial then DIALSTATUS is set to CHANUNAVAIL.

app_mixmonitor
------------------
 * An option 'S' has been added to MixMonitor. If used in combination
   with the r() and/or t() options, if a frame is available to
   write to one of those files but not the other, a frame of silence
   if written to the file that does not have an audio frame. This
   should prevent the two files from "drifting" when mixed after
   the fact.

 * If the 'filename' argument to MixMonitor() ended with '.wav49,'
   Asterisk would silently convert the extension to '.WAV' when
   opening the file for writing. This caused the MIXMONITOR_FILENAME
   variable to reference the wrong file. The MIXMONITOR_FILENAME
   variable will now reflect the name of the file that Asterisk
   actually used instead of the filename that was passed to the
   application.

app_page
------------------
 * The Page application now tolerates empty positions in the supplied
   destination list.  Dialplan can now be simplified by not having
   to check for empty positions in the destination list.

app_voicemail
------------------
 * A feature was added in Asterisk 13.27.0 and 16.4.0 that removed
   lock files from the Asterisk voicemail directory on startup.
   Some users that store their voicemails on network storage devices
   experienced slow startup times due to the relative expense of
   traversing the voicemail directory structure looking for orphaned
   lock files. This feature has now been removed.

   Users who require the lock files to be removed at startup should
   modify their startup scripts to do so before starting the asterisk
   process.

chan_pjsip
------------------
 * A new dialplan function, PJSIP_MOH_PASSTRHOUGH, has been added
   to chan_pjsip. This allows the behaviour of the moh_passthrough
   endpoint option to be read or changed in the dialplan. This
   allows control on a per-call basis.

chan_rtp
------------------
 * The UnicastRTP channel driver provided by chan_rtp now accepts
   "<hostname>:<port>" as an alternative to "<ip_address>:<port>"
   in the destination.  The first AAAA (preferred) or A record
   resolved will be used as the destination.  The lookup is
   synchronous so beware of possible dialplan delays if you specify
   a hostname.

func_curl
------------------
 * A new parameter, httpheader, has been added to CURLOPT function.
   This parameter allows to set custom http headers for subsequent
   calls of CURL function.  Any setting of headers will replace
   the default curl headers (e.g. "Content-type:
   application/x-www-form-urlencoded")

 * A new option, followlocation, can now be enabled with the
   CURLOPT() dialplan function. Setting this will instruct cURL to
   follow 3xx redirects, which it does not by default.

func_jitterbuffer
------------------
 * The JITTERBUFFER dialplan function now has an option to enable
   video synchronization support. When enabled and used with a
   compatible channel driver (chan_sip, chan_pjsip) the video is
   buffered according to the size of the audio jitterbuffer and is
   synchronized to the audio.

func_volume
------------------
 * Accept decimal number as argument.

http
------------------
 * You can now disable the /httpstatus page served by Asterisk's
   built-in HTTP server by setting 'enable_status' to 'no' in
   http.conf.

minmemfree
------------------
 * The 'minmemfree' configuration option now counts memory allocated
   to the filesystem cache as "free" because it is memory that is
   available to the process.

res_ari_channels
------------------
 * When creating a channel in ARI using the create call
   you can now specify dialplan variables to be set as part of the
   same operation.

res_musiconhold
------------------
 * This fix allows a realtime moh class to be unregistered from
   the command line. This is useful when the contents of a directory
   referenced by a realtime moh class have changed.  The realtime
   moh class is then reloaded on the next request and uses the new
   directory contents.

 * A new mode - playlist - has been added to res_musiconhold. This
   mode allows the user to specify the files (or URLs) to play
   explicitly by putting them directly in musiconhold.conf.

res_pjsip
------------------
 * Added a new PJSIP system setting called disable_rport.
   Default is no to keep support working as before.

   If it is false (default) it adds the 'rport' parameter in the
   outgoing request message.  If it is true it does not add the
   'rport' parameter in the outgoing request message.

   This is a system option, but working as a global option.

res_pjsip_endpoint_identifier_ip
------------------
 * In 'type = identify' sections, the addresses specified for the
   'match' clause can now include a port number. For IP addresses,
   the port is provided by including a colon after the address,
   followed by the desired port number. If supplied, the netmask
   should follow the port number. To specify a port for IPv6
   addresses, the address itself must be enclosed in brackets to
   be parsed correctly.

res_pjsip_logger
------------------
 * The PJSIP packet logger now has the following CLI commands:

   pjsip set logger pcap <filename>

   When used this will create a pcap file containing the incoming
   and outgoing SIP packets, in unencrypted form.

   pjsip set logger console <on / off>

   This allows you to toggle logging to console on and off.

   pjsip set logger host <IP/subnet mask> add

   This allows you to add an additional IP address or subnet mask
   to logging, allowing you to log multiple instead of just a single
   IP address or all traffic.

   The normal "pjsip set logger host" CLI command has also been
   expanded to allow subnet masks as well.

res_pjsip_session
------------------
 * When placing an outgoing call to a PJSIP endpoint the intent
   of any requested formats will now be respected. If only an audio
   format is requested (such as ulaw) but the underlying endpoint
   does not support the format the resulting SDP will still only
   contain an audio stream, and not any additional streams such as
   video.

 * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
   have been added to res_pjsip endpoints that specify the preferred
   order of codecs to use between those received/sent in an SDP
   offer and those set in the endpoint configuration.

res_rtp_asterisk
------------------
 * This change include a new cli command 'rtp show settings'

   The command display by general settings of rtp configuration.
   For this point is added the fields: rtpstart, rtpend, dtmftimeout,
   rtpchecksum, strictrtp, learning_min_sequential and icesupport.

 * The blacklist mechanism in res_rtp_asterisk for ICE and STUN
   was converted to an ACL mechanism.

   As such six new options are now available:

   ice_deny
   ice_permit
   ice_acl
   stun_deny
   stun_permit
   stun_acl

   These options have their obvious meanings as used elsewhere.

   Backwards compatibility was maintained by adding {stun,ice}_blacklist
   as aliases for {stun,ice}_deny.

res_sorcery_memory_cache
------------------
 * The SorceryMemoryCacheExpireObject AMI action and CLI
   command allow expiring of a specific object within the sorcery
   memory cache. This is done by removing the object from the cache
   with the expectation that the cache will then re-populate the
   object when it is next needed.

   For full backend caching this does not occur. The cache won't
   repopulate until an entire refresh is done resulting in the
   possibility that objects are missing until that time.

   The AMI action and CLI command will now not allow expiring of
   an object if the cache is configured as a full backend cache.
   Instead you must use either the SorceryMemoryCacheExpire or
   SorceryMemoryCachePopulate AMI actions or their associated CLI
   commands.

taskprocessor.c
------------------
 * Added two new CLI commands to reset stats for taskprocessors.
   You can reset stats for a single, specific taskprocessor ('core
   reset taskprocessor <taskprocessor>'), or you can reset all
   taskprocessors ('core reset taskprocessors'). These commands
   will reset the counter for the number of tasks processed as well
   as the max queue size.

 * Added "like" support for 'core show taskprocessors'. Now you
   can specify a specific set of taskprocessors (or just one) by
   adding the keyword "like" to the above command, followed by your
   search criteria.
@
text
@d1 1
a1 1
# $NetBSD: Makefile,v 1.30 2021/01/17 08:32:40 jnemeth Exp $
d4 1
a4 3
#       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
d6 1
a6 6
DISTNAME=	asterisk-18.4.0
#PKGREVISION=	24
CATEGORIES=	comms net audio
MASTER_SITES=	http://downloads.asterisk.org/pub/telephony/asterisk/
MASTER_SITES+=	http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/
MASTER_SITES+=	http://downloads.asterisk.org/pub/telephony/sounds/releases/
d9 5
d15 2
d20 1
a20 5
EXTRACT_ONLY=	${DISTNAME}.tar.gz
OWNER=		jnemeth@@NetBSD.org
HOMEPAGE=	https://www.asterisk.org/

MAKE_JOBS_SAFE=	NO
d22 1
a22 4
# 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]*
d26 1
a26 1
USE_TOOLS+=		bison gmake perl:run pkg-config tar bash:run
a27 4
REPLACE_BASH+=		contrib/scripts/astversion
REPLACE_BASH+=		contrib/scripts/ast_coredumper
REPLACE_BASH+=		contrib/scripts/ast_logescalator
REPLACE_BASH+=		contrib/scripts/ast_loggrabber
a30 3
REPLACE_PYTHON+=	contrib/scripts/reflocks.py contrib/scripts/refstats.py

CHECK_INTERPRETER_SKIP+=	libdata/asterisk/scripts/refcounter.py
a38 9
.if (${OPSYS} == "NetBSD" && \
    (empty(OS_VERSION:M6.99.[4-9]?) && empty(OS_VERSION:M[7-9].*)))
CONFIGURE_ARGS+=	--without-srtp
.else
CONFIGURE_ARGS+=	--with-srtp
PLIST.srtp=		YES
.include "../../comms/srtp/buildlink3.mk"
.endif

d46 5
a50 4
ASTERISK_USER?=			asterisk
ASTERISK_GROUP?=		asterisk
PKG_GROUPS=			${ASTERISK_GROUP}
PKG_USERS=			${ASTERISK_USER}:${ASTERISK_GROUP}
d52 6
a57 6
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}
a92 39
MAKE_FLAGS+=		OPTIMIZE=-O3

.if !empty(MACHINE_PLATFORM:MSunOS-*-i386)
BUILDLINK_TRANSFORM+=	rm:-march=i386
.endif

PLIST_VARS+=	kqueue
.if exists(/usr/include/sys/event.h)
PLIST.kqueue=	yes
.endif

PLIST_VARS+=	mgcp
# NOSIGPIPE is a temp variable, since PLIST.mgcp MUST remain undefined
# if the grep fails
.for dir in ${COMPILER_INCLUDE_DIRS}
.  if (exists(${dir}/sys/socket.h))
NOSIGPIPE!=	${GREP} SO_NOSIGPIPE ${dir}/sys/socket.h || echo ""
.    if ${NOSIGPIPE} != ""
PLIST.mgcp=	yes
.    endif
.  endif
.endfor

.if ${OPSYS} == "SunOS"
PLIST.mgcp=	yes
.endif

PLIST_VARS+=	timerfd
.if ${OPSYS} == "Linux"
PLIST.timerfd=	yes
.endif

PLIST_VARS+=	unbound
# unbound 1.5 or later is required.`
.if exists(/usr/include/unbound.h) && !empty(OS_VERSION:M8.*)
CONFIGURE_ARGS+=	--without-unbound
.else
PLIST.unbound=	yes
.endif
d97 1
a97 10
DISTFILES+=	asterisk-extra-sounds-en-gsm-1.5.2.tar.gz

# pjproject
PJPROJ_VERSION=	2.10
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}/${DIST_SUBDIR}
d102 6
a107 11
SUBST_FILES.configs=	configs/samples/cli_aliases.conf.sample
SUBST_FILES.configs+=	configs/samples/festival.conf.sample
SUBST_FILES.configs+=	configs/samples/http.conf.sample
SUBST_FILES.configs+=	configs/samples/iax.conf.sample
SUBST_FILES.configs+=	configs/samples/musiconhold.conf.sample
SUBST_FILES.configs+=	configs/samples/osp.conf.sample
SUBST_FILES.configs+=	configs/samples/phoneprov.conf.sample
SUBST_FILES.configs+=	configs/samples/res_config_sqlite.conf.sample
SUBST_FILES.configs+=	configs/samples/sla.conf.sample
SUBST_SED.configs+=	-e 's|doc/|${PREFIX}/share/doc/${PKGBASE}/|'
SUBST_SED.configs+=	-e 's|/etc/asterisk|${ASTETCDIR}|'
d113 7
a119 19
# 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|"

# XXX gross hack, gethostbyname_r on NETBSD is for internal use only
SUBST_CLASSES.NetBSD+=		gethostbyname_r
SUBST_STAGE.gethostbyname_r=	post-configure
SUBST_FILES.gethostbyname_r=	include/asterisk/autoconfig.h
SUBST_SED.gethostbyname_r=	-e "s|^\#define HAVE_GETHOSTBYNAME_R_5 1|\#undef HAVE_GETHOSTBYNAME_R_5|"
SUBST_SED.gethostbyname_r+=	-e "s|^\#define HAVE_GETHOSTBYNAME_R_6 1|\#undef HAVE_GETHOSTBYNAME_R_6|"
a126 1
OWN_DIRS_PERMS+=	${ASTSPOOLDIR}/outgoing ${ASTERISK_USER} ${ASTERISK_GROUP} 0755
a146 1
.if !empty(PKG_OPTIONS:Masterisk-config)
d148 2
a149 3
.  for f in acl.conf adsi.conf agents.conf alarmreceiver.conf alsa.conf	\
	amd.conf app_mysql.conf app_skel.conf ari.conf			\
	ast_debug_tools.conf asterisk.adsi calendar.conf ccss.conf	\
d151 20
a170 23
	cdr_manager.conf cdr_mysql.conf cdr_odbc.conf cdr_pgsql.conf	\
	cdr_sqlite3_custom.conf cdr_syslog.conf cdr_tds.conf cel.conf	\
	cel_custom.conf cel_odbc.conf cel_pgsql.conf			\
	cel_sqlite3_custom.conf cel_tds.conf chan_dahdi.conf		\
	chan_mobile.conf cli.conf cli_aliases.conf cli_permissions.conf	\
	codecs.conf confbridge.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	\
	hep.conf http.conf iax.conf iaxprov.conf indications.conf	\
	logger.conf manager.conf meetme.conf mgcp.conf minivm.conf	\
	misdn.conf modules.conf	motif.conf musiconhold.conf muted.conf	\
	ooh323.conf osp.conf oss.conf phone.conf phoneprov.conf		\
	pjproject.conf pjsip.conf pjsip_notify.conf pjsip_wizard.conf	\
	queuerules.conf queues.conf res_config_mysql.conf		\
	res_config_sqlite.conf res_config_sqlite3.conf			\
	res_corosync.conf res_curl.conf	 res_fax.conf res_ldap.conf	\
	res_odbc.conf res_parking.conf res_pgsql.conf res_pktccops.conf	\
	res_snmp.conf res_stun_monitor.conf resolver_unbound.conf	\
	rtp.conf say.conf sip.conf sip_notify.conf skinny.conf sla.conf	\
	smdi.conf sorcery.conf ss7.timers stasis.conf statsd.conf	\
	stir_shaken.conf telcordia-1.adsi udptl.conf unistim.conf	\
	users.conf voicemail.conf vpb.conf xmpp.conf
d172 1
a172 2
.  endfor
.endif
d183 1
a183 1
	${TAR} xzf ${WRKSRC}/sounds/asterisk-core-sounds-en-gsm-1.6.1.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
d185 1
a185 1
	${TAR} xzf ${DISTDIR}/${DIST_SUBDIR}/asterisk-extra-sounds-en-gsm-1.5.2.tar.gz -C ${DESTDIR}${ASTDATADIR}/sounds/en
d192 5
a196 2
	${INSTALL_DATA} ${WRKSRC}/README.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
	${INSTALL_DATA} ${WRKSRC}/README-SERIOUSLY.bestpractices.md ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
d199 1
a199 1
	${INSTALL_DATA} ${WRKSRC}/doc/IAX2-security.pdf ${DESTDIR}${PREFIX}/share/doc/${PKGBASE}
d202 1
a203 7
.if ${OPSYS} != "Darwin" && ${PKGSRC_COMPILER:Mclang}
.include "../../lang/libBlocksRuntime/buildlink3.mk"
LDFLAGS+= -L${PREFIX}/lib
.endif

.include "../../databases/sqlite3/buildlink3.mk"
.include "../../devel/libuuid/buildlink3.mk"
a204 2
.include "../../lang/python/application.mk"
.include "../../textproc/jansson/buildlink3.mk"
a209 1
.include "../../mk/termcap.buildlink3.mk"
@

