#global _rc 4 #global _beta 5 %if 0%{?fedora} >= 15 %global astvarrundir /run/asterisk %global tmpfilesd 1 %else %global astvarrundir %{_localstatedir}/run/asterisk %global tmpfilesd 0 %endif %if 0%{?fedora} >= 16 %global systemd 1 %else %global systemd 0 %endif Summary: The Open Source PBX Name: asterisk Version: 1.8.12.0 Release: 1%{?_rc:.rc%{_rc}}%{?_beta:.beta%{_beta}}%{?dist} License: GPLv2 Group: Applications/Internet URL: http://www.asterisk.org/ Source0: http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-%{version}%{?_rc:-rc%{_rc}}%{?_beta:-beta%{_beta}}.tar.gz Source1: http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-%{version}%{?_rc:-rc%{_rc}}%{?_beta:-beta%{_beta}}.tar.gz.asc Source2: asterisk-logrotate Source3: menuselect.makedeps Source4: menuselect.makeopts Source5: asterisk.service Source6: asterisk-tmpfiles Patch1: 0001-Modify-init-scripts-for-better-Fedora-compatibilty.patch Patch2: 0002-Modify-modules.conf-so-that-different-voicemail-modu.patch # Submitted upstream: https://issues.asterisk.org/view.php?id=16858 Patch3: 0003-Allow-linking-building-against-an-external-libedit.patch Patch4: 0004-Use-the-library-function-for-loading-command-history.patch Patch5: 0005-Fix-up-some-paths.patch Patch6: 0006-Add-LDAP-schema-that-is-compatible-with-Fedora-Direc.patch Patch7: 0007-Don-t-load-chan_mgcp-and-res_pktccops-because-res_pk.patch #Patch8: 0008-Make-sure-that-AST_ARGS-is-used-consistently-in-Fedo.patch #Patch9: 0009-Use-consistently-in-the-Fedora-init-script.patch #Patch10: 0010-Make-sure-that-the-Fedora-init-script-can-find-the-p.patch # Submitted upstream: https://issues.asterisk.org/jira/browse/ASTERISK-18331 (now merged) #Patch11: 0011-This-fixes-the-inotify-code-to-handle-call-files-bei.patch Patch12: 0012-Fix-two-problems-with-app_sms.patch Patch13: 0013-Remove-blank-lines-to-improve-compat-with-389-Direct.patch BuildRoot: %{_tmppath}/%{name}-%{version}-root-%(%{__id_u} -n) BuildRequires: autoconf BuildRequires: automake # core build requirements BuildRequires: openssl-devel BuildRequires: newt-devel %if 0%{?fedora} <= 8 BuildRequires: libtermcap-devel %endif BuildRequires: ncurses-devel BuildRequires: libcap-devel BuildRequires: gtk2-devel %ifnarch ppc64 BuildRequires: libsrtp-devel %endif %if %{systemd} BuildRequires: systemd-units %endif # for res_http_post %if 0%{?fedora} > 0 BuildRequires: gmime22-devel %endif # for building docs BuildRequires: doxygen BuildRequires: graphviz BuildRequires: graphviz-gd BuildRequires: libxml2-devel BuildRequires: latex2html # for building res_calendar_caldav BuildRequires: neon-devel BuildRequires: libical-devel # for codec_speex BuildRequires: speex-devel >= 1.2 # for format_ogg_vorbis BuildRequires: libogg-devel BuildRequires: libvorbis-devel # codec_gsm BuildRequires: gsm-devel # cli BuildRequires: libedit-devel Requires(pre): %{_sbindir}/useradd Requires(pre): %{_sbindir}/groupadd %if %{systemd} Requires(post): systemd-units Requires(post): systemd-sysv Requires(preun): systemd-units Requires(postun): systemd-units %else Requires(post): /sbin/chkconfig Requires(preun): /sbin/chkconfig Requires(preun): /sbin/service %endif # asterisk-conference package removed since patch no longer compiles Obsoletes: asterisk-conference <= 1.6.0-0.14.beta9 Obsoletes: asterisk-mobile <= 1.6.1-0.23.rc1 Obsoletes: asterisk-firmware <= 1.6.2.0-0.2.rc1 # chan_usbradio was removed in 1.8.12.0 Obsoletes: asterisk-usbradio <= 1.8.11.1-1 %description Asterisk is a complete PBX in software. It runs on Linux and 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. %if 0%{?fedora} > 0 %package ais Summary: Modules for Asterisk that use OpenAIS Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: openais-devel %description ais Modules for Asterisk that use OpenAIS. %endif %package alsa Summary: Modules for Asterisk that use Alsa sound drivers Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: alsa-lib-devel %description alsa Modules for Asterisk that use Alsa sound drivers. %package apidoc Summary: API documentation for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description apidoc API documentation for Asterisk. %package calendar Summary: Calendar applications for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description calendar Calendar applications for Asterisk. %package curl Summary: Modules for Asterisk that use cURL Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: curl-devel %description curl Modules for Asterisk that use cURL. %package dahdi Summary: Modules for Asterisk that use DAHDI Group: Applications/Internet Requires: asterisk = %{version}-%{release} Requires: dahdi-tools >= 2.0.0 Requires(pre): %{_sbindir}/usermod BuildRequires: dahdi-tools-devel >= 2.0.0 BuildRequires: dahdi-tools-libs >= 2.0.0 BuildRequires: libpri-devel >= 1.4.12 BuildRequires: libss7-devel >= 1.0.1 Obsoletes: asterisk-zaptel <= 1.6.0-0.22.beta9 Provides: asterisk-zaptel = %{version}-%{release} %description dahdi Modules for Asterisk that use DAHDI. %package devel Summary: Development files for Asterisk Group: Development/Libraries Requires: asterisk = %{version}-%{release} %description devel Development files for Asterisk. %package fax Summary: FAX applications for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: spandsp-devel >= 0.0.5-0.1.pre4 %description fax FAX applications for Asterisk %package festival Summary: Festival application for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} Requires: festival %description festival Application for the Asterisk PBX that uses Festival to convert text to speech. %if 0%{?fedora} %package ices Summary: Stream audio from Asterisk to an IceCast server Group: Applications/Internet Requires: asterisk = %{version}-%{release} Requires: ices Obsoletes: asterisk < 1.4.18-1 Conflicts: asterisk < 1.4.18-1 %description ices Stream audio from Asterisk to an IceCast server. %endif %package jabber Summary: Jabber/XMPP resources for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: iksemel-devel %description jabber Jabber/XMPP resources for Asterisk. %package jack Summary: JACK resources for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: jack-audio-connection-kit-devel BuildRequires: libresample-devel %description jack JACK resources for Asterisk. %package lua Summary: Lua resources for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: lua-devel %description lua Lua resources for Asterisk. %package ldap Summary: LDAP resources for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: openldap-devel %description ldap LDAP resources for Asterisk. %if 0%{?rhel} <= 5 || 0%{?fedora} %package ldap-389 Summary: LDAP resources for Asterisk and the 389 Directory Server Group: Applications/Internet Requires: asterisk = %{version}-%{release} Requires: asterisk-ldap = %{version}-%{release} Requires: 389-ds-base Obsoletes: asterisk-ldap-fds < 1.8.4.4-2 Conflicts: asterisk-ldap-fds < 1.8.4.4-2 %description ldap-389 LDAP resources for Asterisk and the 389 Directory Server. %endif %package misdn Summary: mISDN channel for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} Requires(pre): %{_sbindir}/usermod BuildRequires: mISDN-devel %description misdn mISDN channel for Asterisk. %package mobile Summary: Mobile (BlueTooth) channel for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} Requires(pre): %{_sbindir}/usermod BuildRequires: bluez-libs-devel %description mobile Mobile (BlueTooth) channel for Asterisk. %package minivm Summary: MiniVM applicaton for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description minivm MiniVM application for Asterisk. %package mysql Summary: Applications for Asterisk that use MySQL Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: mysql-devel %description mysql Applications for Asterisk that use MySQL. %package odbc Summary: Applications for Asterisk that use ODBC (except voicemail) Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: libtool-ltdl-devel BuildRequires: unixODBC-devel %description odbc Applications for Asterisk that use ODBC (except voicemail) %package ooh323 Summary: H.323 channel for Asterisk using the Objective Systems Open H.323 for C library Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: libtool-ltdl-devel BuildRequires: unixODBC-devel %description ooh323 H.323 channel for Asterisk using the Objective Systems Open H.323 for C library. %package oss Summary: Modules for Asterisk that use OSS sound drivers Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description oss Modules for Asterisk that use OSS sound drivers. %package portaudio Summary: Modules for Asterisk that use the portaudio library Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: portaudio-devel >= 19 %description portaudio Modules for Asterisk that use the portaudio library. %package postgresql Summary: Applications for Asterisk that use PostgreSQL Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: postgresql-devel %description postgresql Applications for Asterisk that use PostgreSQL. %package radius Summary: Applications for Asterisk that use RADIUS Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: radiusclient-ng-devel %description radius Applications for Asterisk that use RADIUS. %package skinny Summary: Modules for Asterisk that support the SCCP/Skinny protocol Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description skinny Modules for Asterisk that support the SCCP/Skinny protocol. %package snmp Summary: Module that enables SNMP monitoring of Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: net-snmp-devel BuildRequires: lm_sensors-devel # This subpackage depends on perl-libs, this Requires tracks versioning. Requires: perl(:MODULE_COMPAT_%(eval "`%{__perl} -V:version`"; echo $version)) %description snmp Module that enables SNMP monitoring of Asterisk. %package sqlite Summary: Sqlite modules for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: sqlite-devel %description sqlite Sqlite modules for Asterisk. %package tds Summary: Modules for Asterisk that use FreeTDS Group: Applications/Internet Requires: asterisk = %{version}-%{release} BuildRequires: freetds-devel %description tds Modules for Asterisk that use FreeTDS. %package unistim Summary: Unistim channel for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description unistim Unistim channel for Asterisk %package voicemail Summary: Common Voicemail Modules for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} Requires: asterisk-voicemail-implementation = %{version}-%{release} Requires: /usr/bin/sox Requires: /usr/sbin/sendmail %description voicemail Common Voicemail Modules for Asterisk. %if 0%{?fedora} > 0 %package voicemail-imap Summary: Store voicemail on an IMAP server Group: Applications/Internet Requires: asterisk = %{version}-%{release} Requires: asterisk-voicemail = %{version}-%{release} Provides: asterisk-voicemail-implementation = %{version}-%{release} BuildRequires: uw-imap-devel %description voicemail-imap Voicemail implementation for Asterisk that stores voicemail on an IMAP server. %endif %package voicemail-odbc Summary: Store voicemail in a database using ODBC Group: Applications/Internet Requires: asterisk = %{version}-%{release} Requires: asterisk-voicemail = %{version}-%{release} Provides: asterisk-voicemail-implementation = %{version}-%{release} %description voicemail-odbc Voicemail implementation for Asterisk that uses ODBC to store voicemail in a database. %package voicemail-plain Summary: Store voicemail on the local filesystem Group: Applications/Internet Requires: asterisk = %{version}-%{release} Requires: asterisk-voicemail = %{version}-%{release} Provides: asterisk-voicemail-implementation = %{version}-%{release} %description voicemail-plain Voicemail implementation for Asterisk that stores voicemail on the local filesystem. %prep %setup0 -q -n asterisk-%{version}%{?_rc:-rc%{_rc}}%{?_beta:-beta%{_beta}} %patch1 -p1 %patch2 -p1 %patch3 -p1 %patch4 -p1 %patch5 -p1 %patch6 -p1 %patch7 -p1 #patch8 -p1 #patch9 -p1 #patch10 -p1 #patch11 -p1 %patch12 -p1 %patch13 -p1 cp %{S:3} menuselect.makedeps cp %{S:4} menuselect.makeopts # Fixup makefile so sound archives aren't downloaded/installed %{__perl} -pi -e 's/^all:.*$/all:/' sounds/Makefile %{__perl} -pi -e 's/^install:.*$/install:/' sounds/Makefile # convert comments in one file to UTF-8 mv main/fskmodem.c main/fskmodem.c.old iconv -f iso-8859-1 -t utf-8 -o main/fskmodem.c main/fskmodem.c.old touch -r main/fskmodem.c.old main/fskmodem.c rm main/fskmodem.c.old chmod -x contrib/scripts/dbsep.cgi # no openais-devel available for el6 %if 0%{?rhel} == 6 %{__perl} -pi -e 's/^MENUSELECT_RES=(.*)$/MENUSELECT_RES=\1 res_ais res_http_post/g' menuselect.makeopts %endif %ifarch ppc64 %{__perl} -pi -e 's/^MENUSELECT_RES=(.*)$/MENUSELECT_RES=\1 res_srtp/g' menuselect.makeopts %endif %if 0%{?rhel} == 5 # Get the autoconf scripts working with 2.59 %{__perl} -pi -e 's/AC_PREREQ\(2\.60\)/AC_PREREQ\(2\.59\)/g' configure.ac %{__perl} -pi -e 's/AC_USE_SYSTEM_EXTENSIONS/AC_GNU_SOURCE/g' configure.ac %{__perl} -pi -e 's/AST_PROG_SED/SED=sed/g' autoconf/ast_prog_ld.m4 # kernel/glibc in RHEL5 does not support the timerfd %{__perl} -pi -e 's/^MENUSELECT_RES=(.*)$/MENUSELECT_RES=\1 res_timing_timerfd/g' menuselect.makeopts %endif %build %define optflags %(rpm --eval %%{optflags}) -Werror-implicit-function-declaration aclocal -I autoconf autoconf autoheader pushd menuselect/mxml %configure popd pushd menuselect %configure popd %if 0%{?fedora} > 0 %ifnarch ppc64 %configure --with-imap=system --with-gsm=/usr --with-libedit=yes --with-srtp %else %configure --with-imap=system --with-gsm=/usr --with-libedit=yes %endif %else %ifnarch ppc64 %configure --with-gsm=/usr --with-libedit=yes --with-gmime=no --with-srtp %else %configure --with-gsm=/usr --with-libedit=yes --with-gmime=no %endif %endif make menuselect-tree %{__perl} -n -i -e 'print unless /openr2/i' menuselect-tree ASTCFLAGS="%{optflags}" make DEBUG= OPTIMIZE= ASTVARRUNDIR=%{astvarrundir} ASTDATADIR=%{_datadir}/asterisk ASTVARLIBDIR=%{_datadir}/asterisk ASTDBDIR=%{_localstatedir}/spool/asterisk NOISY_BUILD=1 rm apps/app_voicemail.o apps/app_directory.o mv apps/app_voicemail.so apps/app_voicemail_plain.so mv apps/app_directory.so apps/app_directory_plain.so %if 0%{?fedora} > 0 %{__sed} -i -e 's/^MENUSELECT_OPTS_app_voicemail=.*$/MENUSELECT_OPTS_app_voicemail=IMAP_STORAGE/' menuselect.makeopts ASTCFLAGS="%{optflags}" make DEBUG= OPTIMIZE= ASTVARRUNDIR=%{astvarrundir} ASTDATADIR=%{_datadir}/asterisk ASTVARLIBDIR=%{_datadir}/asterisk ASTDBDIR=%{_localstatedir}/spool/asterisk NOISY_BUILD=1 rm apps/app_voicemail.o apps/app_directory.o mv apps/app_voicemail.so apps/app_voicemail_imap.so mv apps/app_directory.so apps/app_directory_imap.so %endif %{__sed} -i -e 's/^MENUSELECT_OPTS_app_voicemail=.*$/MENUSELECT_OPTS_app_voicemail=ODBC_STORAGE/' menuselect.makeopts ASTCFLAGS="%{optflags}" make DEBUG= OPTIMIZE= ASTVARRUNDIR=%{astvarrundir} ASTDATADIR=%{_datadir}/asterisk ASTVARLIBDIR=%{_datadir}/asterisk ASTDBDIR=%{_localstatedir}/spool/asterisk NOISY_BUILD=1 rm apps/app_voicemail.o apps/app_directory.o mv apps/app_voicemail.so apps/app_voicemail_odbc.so mv apps/app_directory.so apps/app_directory_odbc.so # so that these modules don't get built again during the install phase touch apps/app_voicemail.o apps/app_directory.o touch apps/app_voicemail.so apps/app_directory.so ASTCFLAGS="%{optflags}" make progdocs DEBUG= OPTIMIZE= ASTVARRUNDIR=%{astvarrundir} ASTDATADIR=%{_datadir}/asterisk ASTVARLIBDIR=%{_datadir}/asterisk ASTDBDIR=%{_localstatedir}/spool/asterisk NOISY_BUILD=1 # fix dates so that we don't get multilib conflicts find doc/api/html -type f -print0 | xargs --null touch -r ChangeLog cd doc/tex && ASTCFLAGS="%{optflags}" make html %install rm -rf %{buildroot} ASTCFLAGS="%{optflags}" make install DEBUG= OPTIMIZE= DESTDIR=%{buildroot} ASTVARRUNDIR=%{astvarrundir} ASTDATADIR=%{_datadir}/asterisk ASTVARLIBDIR=%{_datadir}/asterisk ASTDBDIR=%{_localstatedir}/spool/asterisk NOISY_BUILD=1 ASTCFLAGS="%{optflags}" make samples DEBUG= OPTIMIZE= DESTDIR=%{buildroot} ASTVARRUNDIR=%{astvarrundir} ASTDATADIR=%{_datadir}/asterisk ASTVARLIBDIR=%{_datadir}/asterisk ASTDBDIR=%{_localstatedir}/spool/asterisk NOISY_BUILD=1 %if %{systemd} install -D -p -m 0644 %{SOURCE5} %{buildroot}%{_unitdir}/asterisk.service rm -f %{buildroot}%{_sbindir}/safe_asterisk %else install -D -p -m 0755 contrib/init.d/rc.redhat.asterisk %{buildroot}%{_initrddir}/asterisk install -D -p -m 0644 contrib/sysconfig/asterisk %{buildroot}%{_sysconfdir}/sysconfig/asterisk %endif install -D -p -m 0644 contrib/scripts/99asterisk.ldif %{buildroot}%{_sysconfdir}/dirsrv/schema/99asterisk.ldif install -D -p -m 0644 %{S:2} %{buildroot}%{_sysconfdir}/logrotate.d/asterisk #install -D -p -m 0644 doc/asterisk-mib.txt %{buildroot}%{_datadir}/snmp/mibs/ASTERISK-MIB.txt #install -D -p -m 0644 doc/digium-mib.txt %{buildroot}%{_datadir}/snmp/mibs/DIGIUM-MIB.txt rm %{buildroot}%{_libdir}/asterisk/modules/app_directory.so rm %{buildroot}%{_libdir}/asterisk/modules/app_voicemail.so %if 0%{?fedora} > 0 install -D -p -m 0755 apps/app_directory_imap.so %{buildroot}%{_libdir}/asterisk/modules/app_directory_imap.so install -D -p -m 0755 apps/app_voicemail_imap.so %{buildroot}%{_libdir}/asterisk/modules/app_voicemail_imap.so %endif install -D -p -m 0755 apps/app_directory_odbc.so %{buildroot}%{_libdir}/asterisk/modules/app_directory_odbc.so install -D -p -m 0755 apps/app_voicemail_odbc.so %{buildroot}%{_libdir}/asterisk/modules/app_voicemail_odbc.so install -D -p -m 0755 apps/app_directory_plain.so %{buildroot}%{_libdir}/asterisk/modules/app_directory_plain.so install -D -p -m 0755 apps/app_voicemail_plain.so %{buildroot}%{_libdir}/asterisk/modules/app_voicemail_plain.so # create some directories that need to be packaged mkdir -p %{buildroot}%{_datadir}/asterisk/moh mkdir -p %{buildroot}%{_datadir}/asterisk/sounds mkdir -p %{buildroot}%{_localstatedir}/lib/asterisk mkdir -p %{buildroot}%{_localstatedir}/log/asterisk/cdr-custom mkdir -p %{buildroot}%{_localstatedir}/spool/asterisk/festival mkdir -p %{buildroot}%{_localstatedir}/spool/asterisk/monitor mkdir -p %{buildroot}%{_localstatedir}/spool/asterisk/outgoing mkdir -p %{buildroot}%{_localstatedir}/spool/asterisk/uploads # We're not going to package any of the sample AGI scripts rm -f %{buildroot}%{_datadir}/asterisk/agi-bin/* # Don't package the sample voicemail user rm -rf %{buildroot}%{_localstatedir}/spool/asterisk/voicemail/default # Don't package example phone provision configs rm -rf %{buildroot}%{_datadir}/asterisk/phoneprov/* # these are compiled with -O0 and thus include unfortified code. rm -rf %{buildroot}%{_sbindir}/hashtest rm -rf %{buildroot}%{_sbindir}/hashtest2 find doc/api/html -name \*.map -size 0 -delete %if 0%{?fedora} == 0 rm -f %{buildroot}%{_sysconfdir}/asterisk/ais.conf %endif #rhel6 doesnt have 389 available, nor ices %if 0%{?rhel} == 6 rm -rf %{buildroot}%{_sysconfdir}/dirsrv/schema/99asterisk.ldif rm -rf %{buildroot}%{_libdir}/asterisk/modules/app_ices.so %endif %if %{tmpfilesd} install -D -p -m 0644 %{SOURCE6} %{buildroot}/usr/lib/tmpfiles.d/asterisk.conf %endif rm %{buildroot}%{_sysconfdir}/asterisk/usbradio.conf %clean rm -rf %{buildroot} %pre %{_sbindir}/groupadd -r asterisk &>/dev/null || : %{_sbindir}/useradd -r -s /sbin/nologin -d /var/lib/asterisk -M \ -c 'Asterisk User' -g asterisk asterisk &>/dev/null || : %post %if %{systemd} if [ $1 -eq 1 ] ; then /bin/systemctl daemon-reload >/dev/null 2>&1 || : fi %else /sbin/chkconfig --add asterisk %endif %preun %if %{systemd} if [ "$1" -eq "0" ]; then # Package removal, not upgrade /bin/systemctl --no-reload disable asterisk.service > /dev/null 2>&1 || : /bin/systemctl stop asterisk.service > /dev/null 2>&1 || : fi %else if [ "$1" -eq "0" ]; then # Package removal, not upgrade /sbin/service asterisk stop > /dev/null 2>&1 || : /sbin/chkconfig --del asterisk fi %endif %if %{systemd} %postun /bin/systemctl daemon-reload >/dev/null 2>&1 || : if [ $1 -ge 1 ] ; then # Package upgrade, not uninstall /bin/systemctl try-restart asterisk.service >/dev/null 2>&1 || : fi %triggerun -- asterisk < 1.8.2.4-2 # Save the current service runlevel info # User must manually run systemd-sysv-convert --apply asterisk # to migrate them to systemd targets /usr/bin/systemd-sysv-convert --save asterisk >/dev/null 2>&1 ||: # Run these because the SysV package being removed won't do them /sbin/chkconfig --del asterisk >/dev/null 2>&1 || : /bin/systemctl try-restart asterisk.service >/dev/null 2>&1 || : %endif %pre dahdi %{_sbindir}/usermod -a -G dahdi asterisk %pre misdn %{_sbindir}/usermod -a -G misdn asterisk %files %defattr(-,root,root,-) %doc README* *.txt ChangeLog BUGS CREDITS configs %doc doc/asterisk.sgml #doc doc/backtrace.txt #doc doc/callfiles.txt #doc doc/externalivr.txt #doc doc/macroexclusive.txt #doc doc/manager_1_1.txt #doc doc/modules.txt #doc doc/PEERING #doc doc/queue.txt #doc doc/rtp-packetization.txt #doc doc/siptls.txt #doc doc/smdi.txt #doc doc/sms.txt #doc doc/speechrec.txt #doc doc/ss7.txt #doc doc/video.txt %if %{systemd} %{_unitdir}/asterisk.service %else %{_initrddir}/asterisk %config(noreplace) %{_sysconfdir}/sysconfig/asterisk %endif %dir %{_libdir}/asterisk %dir %{_libdir}/asterisk/modules %{_libdir}/asterisk/modules/app_adsiprog.so %{_libdir}/asterisk/modules/app_alarmreceiver.so %{_libdir}/asterisk/modules/app_amd.so %{_libdir}/asterisk/modules/app_authenticate.so %{_libdir}/asterisk/modules/app_cdr.so %{_libdir}/asterisk/modules/app_celgenuserevent.so %{_libdir}/asterisk/modules/app_chanisavail.so %{_libdir}/asterisk/modules/app_channelredirect.so %{_libdir}/asterisk/modules/app_chanspy.so %{_libdir}/asterisk/modules/app_confbridge.so %{_libdir}/asterisk/modules/app_controlplayback.so %{_libdir}/asterisk/modules/app_db.so %{_libdir}/asterisk/modules/app_dial.so %{_libdir}/asterisk/modules/app_dictate.so %{_libdir}/asterisk/modules/app_directed_pickup.so %{_libdir}/asterisk/modules/app_disa.so %{_libdir}/asterisk/modules/app_dumpchan.so %{_libdir}/asterisk/modules/app_echo.so %{_libdir}/asterisk/modules/app_exec.so %{_libdir}/asterisk/modules/app_externalivr.so %{_libdir}/asterisk/modules/app_followme.so %{_libdir}/asterisk/modules/app_forkcdr.so %{_libdir}/asterisk/modules/app_getcpeid.so %{_libdir}/asterisk/modules/app_image.so %{_libdir}/asterisk/modules/app_macro.so %{_libdir}/asterisk/modules/app_milliwatt.so %{_libdir}/asterisk/modules/app_mixmonitor.so %{_libdir}/asterisk/modules/app_morsecode.so %{_libdir}/asterisk/modules/app_nbscat.so %{_libdir}/asterisk/modules/app_originate.so %{_libdir}/asterisk/modules/app_parkandannounce.so %{_libdir}/asterisk/modules/app_playback.so %{_libdir}/asterisk/modules/app_playtones.so %{_libdir}/asterisk/modules/app_privacy.so %{_libdir}/asterisk/modules/app_queue.so %{_libdir}/asterisk/modules/app_readexten.so %{_libdir}/asterisk/modules/app_readfile.so %{_libdir}/asterisk/modules/app_read.so %{_libdir}/asterisk/modules/app_record.so %{_libdir}/asterisk/modules/app_saycounted.so %{_libdir}/asterisk/modules/app_saycountpl.so %{_libdir}/asterisk/modules/app_sayunixtime.so %{_libdir}/asterisk/modules/app_senddtmf.so %{_libdir}/asterisk/modules/app_sendtext.so %{_libdir}/asterisk/modules/app_setcallerid.so %{_libdir}/asterisk/modules/app_sms.so %{_libdir}/asterisk/modules/app_softhangup.so %{_libdir}/asterisk/modules/app_speech_utils.so %{_libdir}/asterisk/modules/app_stack.so %{_libdir}/asterisk/modules/app_system.so %{_libdir}/asterisk/modules/app_talkdetect.so %{_libdir}/asterisk/modules/app_test.so %{_libdir}/asterisk/modules/app_transfer.so %{_libdir}/asterisk/modules/app_url.so %{_libdir}/asterisk/modules/app_userevent.so %{_libdir}/asterisk/modules/app_verbose.so %{_libdir}/asterisk/modules/app_waitforring.so %{_libdir}/asterisk/modules/app_waitforsilence.so %{_libdir}/asterisk/modules/app_waituntil.so %{_libdir}/asterisk/modules/app_while.so %{_libdir}/asterisk/modules/app_zapateller.so %{_libdir}/asterisk/modules/bridge_builtin_features.so %{_libdir}/asterisk/modules/bridge_multiplexed.so %{_libdir}/asterisk/modules/bridge_simple.so %{_libdir}/asterisk/modules/bridge_softmix.so %{_libdir}/asterisk/modules/cdr_csv.so %{_libdir}/asterisk/modules/cdr_custom.so %{_libdir}/asterisk/modules/cdr_manager.so %{_libdir}/asterisk/modules/cdr_syslog.so %{_libdir}/asterisk/modules/cel_custom.so %{_libdir}/asterisk/modules/cel_manager.so %{_libdir}/asterisk/modules/chan_agent.so %{_libdir}/asterisk/modules/chan_bridge.so %{_libdir}/asterisk/modules/chan_iax2.so %{_libdir}/asterisk/modules/chan_local.so %{_libdir}/asterisk/modules/chan_mgcp.so %{_libdir}/asterisk/modules/chan_multicast_rtp.so %{_libdir}/asterisk/modules/chan_phone.so %{_libdir}/asterisk/modules/chan_sip.so %{_libdir}/asterisk/modules/codec_adpcm.so %{_libdir}/asterisk/modules/codec_alaw.so %{_libdir}/asterisk/modules/codec_a_mu.so %{_libdir}/asterisk/modules/codec_g722.so %{_libdir}/asterisk/modules/codec_g726.so %{_libdir}/asterisk/modules/codec_gsm.so %{_libdir}/asterisk/modules/codec_lpc10.so %{_libdir}/asterisk/modules/codec_resample.so %{_libdir}/asterisk/modules/codec_speex.so %{_libdir}/asterisk/modules/codec_ulaw.so %{_libdir}/asterisk/modules/format_g719.so %{_libdir}/asterisk/modules/format_g723.so %{_libdir}/asterisk/modules/format_g726.so %{_libdir}/asterisk/modules/format_g729.so %{_libdir}/asterisk/modules/format_gsm.so %{_libdir}/asterisk/modules/format_h263.so %{_libdir}/asterisk/modules/format_h264.so %{_libdir}/asterisk/modules/format_jpeg.so %{_libdir}/asterisk/modules/format_ogg_vorbis.so %{_libdir}/asterisk/modules/format_pcm.so %{_libdir}/asterisk/modules/format_siren14.so %{_libdir}/asterisk/modules/format_siren7.so %{_libdir}/asterisk/modules/format_sln.so %{_libdir}/asterisk/modules/format_sln16.so %{_libdir}/asterisk/modules/format_vox.so %{_libdir}/asterisk/modules/format_wav_gsm.so %{_libdir}/asterisk/modules/format_wav.so %{_libdir}/asterisk/modules/func_aes.so %{_libdir}/asterisk/modules/func_audiohookinherit.so %{_libdir}/asterisk/modules/func_base64.so %{_libdir}/asterisk/modules/func_blacklist.so %{_libdir}/asterisk/modules/func_callcompletion.so %{_libdir}/asterisk/modules/func_callerid.so %{_libdir}/asterisk/modules/func_cdr.so %{_libdir}/asterisk/modules/func_channel.so %{_libdir}/asterisk/modules/func_config.so %{_libdir}/asterisk/modules/func_cut.so %{_libdir}/asterisk/modules/func_db.so %{_libdir}/asterisk/modules/func_devstate.so %{_libdir}/asterisk/modules/func_dialgroup.so %{_libdir}/asterisk/modules/func_dialplan.so %{_libdir}/asterisk/modules/func_enum.so %{_libdir}/asterisk/modules/func_env.so %{_libdir}/asterisk/modules/func_extstate.so %{_libdir}/asterisk/modules/func_frame_trace.so %{_libdir}/asterisk/modules/func_global.so %{_libdir}/asterisk/modules/func_groupcount.so %{_libdir}/asterisk/modules/func_iconv.so %{_libdir}/asterisk/modules/func_lock.so %{_libdir}/asterisk/modules/func_logic.so %{_libdir}/asterisk/modules/func_math.so %{_libdir}/asterisk/modules/func_md5.so %{_libdir}/asterisk/modules/func_module.so %{_libdir}/asterisk/modules/func_pitchshift.so %{_libdir}/asterisk/modules/func_rand.so %{_libdir}/asterisk/modules/func_realtime.so %{_libdir}/asterisk/modules/func_sha1.so %{_libdir}/asterisk/modules/func_shell.so %{_libdir}/asterisk/modules/func_speex.so %{_libdir}/asterisk/modules/func_sprintf.so %{_libdir}/asterisk/modules/func_srv.so %{_libdir}/asterisk/modules/func_strings.so %{_libdir}/asterisk/modules/func_sysinfo.so %{_libdir}/asterisk/modules/func_timeout.so %{_libdir}/asterisk/modules/func_uri.so %{_libdir}/asterisk/modules/func_version.so %{_libdir}/asterisk/modules/func_volume.so %{_libdir}/asterisk/modules/pbx_ael.so %{_libdir}/asterisk/modules/pbx_config.so %{_libdir}/asterisk/modules/pbx_dundi.so %{_libdir}/asterisk/modules/pbx_loopback.so %{_libdir}/asterisk/modules/pbx_realtime.so %{_libdir}/asterisk/modules/pbx_spool.so %{_libdir}/asterisk/modules/res_adsi.so %{_libdir}/asterisk/modules/res_ael_share.so %{_libdir}/asterisk/modules/res_agi.so %{_libdir}/asterisk/modules/res_clialiases.so %{_libdir}/asterisk/modules/res_clioriginate.so %{_libdir}/asterisk/modules/res_convert.so %{_libdir}/asterisk/modules/res_crypto.so %if 0%{?fedora} > 0 %{_libdir}/asterisk/modules/res_http_post.so %endif %{_libdir}/asterisk/modules/res_limit.so %{_libdir}/asterisk/modules/res_monitor.so %{_libdir}/asterisk/modules/res_musiconhold.so %{_libdir}/asterisk/modules/res_mutestream.so %{_libdir}/asterisk/modules/res_phoneprov.so %{_libdir}/asterisk/modules/res_pktccops.so %{_libdir}/asterisk/modules/res_realtime.so %{_libdir}/asterisk/modules/res_rtp_asterisk.so %{_libdir}/asterisk/modules/res_rtp_multicast.so %{_libdir}/asterisk/modules/res_security_log.so %{_libdir}/asterisk/modules/res_smdi.so %{_libdir}/asterisk/modules/res_speech.so %ifnarch ppc64 %{_libdir}/asterisk/modules/res_srtp.so %endif %{_libdir}/asterisk/modules/res_stun_monitor.so %{_libdir}/asterisk/modules/res_timing_pthread.so %if 0%{?fedora} > 0 || 0%{?rhel} >= 6 %{_libdir}/asterisk/modules/res_timing_timerfd.so %endif %{_sbindir}/aelparse %{_sbindir}/astcanary %{_sbindir}/asterisk %{_sbindir}/astgenkey %{_sbindir}/astman %{_sbindir}/autosupport %{_sbindir}/conf2ael %{_sbindir}/muted %{_sbindir}/rasterisk %{_sbindir}/refcounter %if !%{systemd} %{_sbindir}/safe_asterisk %endif %{_sbindir}/smsq %{_sbindir}/stereorize %{_sbindir}/streamplayer %{_mandir}/man8/asterisk.8* %{_mandir}/man8/astgenkey.8* %{_mandir}/man8/autosupport.8* %{_mandir}/man8/safe_asterisk.8* %attr(0750,asterisk,asterisk) %dir %{_sysconfdir}/asterisk %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/adsi.conf #%attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/adtranvofr.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/agents.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/alarmreceiver.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/amd.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/asterisk.adsi %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/asterisk.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/ccss.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cdr.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cdr_custom.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cdr_manager.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cdr_syslog.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cel.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cel_custom.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cli.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cli_aliases.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cli_permissions.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/codecs.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/dnsmgr.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/dsp.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/dundi.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/enum.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/extconfig.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/extensions.ael %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/extensions.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/features.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/followme.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/h323.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/http.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/iax.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/iaxprov.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/indications.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/logger.conf %attr(0600,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/manager.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/mgcp.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/modules.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/musiconhold.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/muted.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/osp.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/phone.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/phoneprov.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/queuerules.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/queues.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_pktccops.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_stun_monitor.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/rpt.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/rtp.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/say.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/sip.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/sip_notify.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/sla.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/smdi.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/telcordia-1.adsi %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/udptl.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/users.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/vpb.conf %config(noreplace) %{_sysconfdir}/logrotate.d/asterisk %dir %{_datadir}/asterisk %dir %{_datadir}/asterisk/agi-bin %{_datadir}/asterisk/documentation %dir %{_datadir}/asterisk/firmware %dir %{_datadir}/asterisk/firmware/iax %{_datadir}/asterisk/images %attr(0750,asterisk,asterisk) %{_datadir}/asterisk/keys %{_datadir}/asterisk/phoneprov %{_datadir}/asterisk/static-http %dir %{_datadir}/asterisk/moh %dir %{_datadir}/asterisk/sounds %attr(0750,asterisk,asterisk) %dir %{_localstatedir}/lib/asterisk %attr(0750,asterisk,asterisk) %dir %{_localstatedir}/log/asterisk %attr(0750,asterisk,asterisk) %dir %{_localstatedir}/log/asterisk/cdr-csv %attr(0750,asterisk,asterisk) %dir %{_localstatedir}/log/asterisk/cdr-custom %attr(0750,asterisk,asterisk) %dir %{_localstatedir}/spool/asterisk %attr(0770,asterisk,asterisk) %dir %{_localstatedir}/spool/asterisk/monitor %attr(0770,asterisk,asterisk) %dir %{_localstatedir}/spool/asterisk/outgoing %attr(0750,asterisk,asterisk) %dir %{_localstatedir}/spool/asterisk/tmp %attr(0750,asterisk,asterisk) %dir %{_localstatedir}/spool/asterisk/uploads %attr(0750,asterisk,asterisk) %dir %{_localstatedir}/spool/asterisk/voicemail %if %{tmpfilesd} %attr(0644,root,root) /usr/lib/tmpfiles.d/asterisk.conf %ghost %attr(0755,asterisk,asterisk) %dir %{astvarrundir} %else %attr(0755,asterisk,asterisk) %dir %{astvarrundir} %endif %if 0%{?fedora} > 0 %files ais %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/ais.conf %{_libdir}/asterisk/modules/res_ais.so %endif %files alsa %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/alsa.conf %{_libdir}/asterisk/modules/chan_alsa.so %files apidoc %defattr(-,root,root,-) %doc doc/api/html/* %files calendar %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/calendar.conf %{_libdir}/asterisk/modules/res_calendar.so %{_libdir}/asterisk/modules/res_calendar_caldav.so %{_libdir}/asterisk/modules/res_calendar_ews.so %{_libdir}/asterisk/modules/res_calendar_exchange.so %{_libdir}/asterisk/modules/res_calendar_icalendar.so %files curl %defattr(-,root,root,-) %doc contrib/scripts/dbsep.cgi %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/dbsep.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_curl.conf %{_libdir}/asterisk/modules/func_curl.so %{_libdir}/asterisk/modules/res_config_curl.so %{_libdir}/asterisk/modules/res_curl.so %files dahdi %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/meetme.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/chan_dahdi.conf %{_libdir}/asterisk/modules/app_flash.so %{_libdir}/asterisk/modules/app_meetme.so %{_libdir}/asterisk/modules/app_page.so %{_libdir}/asterisk/modules/app_dahdibarge.so %{_libdir}/asterisk/modules/app_dahdiras.so #%{_libdir}/asterisk/modules/app_dahdiscan.so %{_libdir}/asterisk/modules/chan_dahdi.so %{_libdir}/asterisk/modules/codec_dahdi.so %{_libdir}/asterisk/modules/res_timing_dahdi.so %files devel %defattr(-,root,root,-) #doc doc/CODING-GUIDELINES #doc doc/datastores.txt #doc doc/modules.txt #doc doc/valgrind.txt %dir %{_includedir}/asterisk %dir %{_includedir}/asterisk/doxygen %{_includedir}/asterisk.h %{_includedir}/asterisk/*.h %{_includedir}/asterisk/doxygen/*.h %files fax %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_fax.conf %{_libdir}/asterisk/modules/res_fax.so %{_libdir}/asterisk/modules/res_fax_spandsp.so %files festival %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/festival.conf %attr(0750,asterisk,asterisk) %dir %{_localstatedir}/spool/asterisk/festival %{_libdir}/asterisk/modules/app_festival.so %if 0%{?fedora} %files ices %defattr(-,root,root,-) %doc contrib/asterisk-ices.xml %{_libdir}/asterisk/modules/app_ices.so %endif %files jabber %defattr(-,root,root,-) #doc doc/jabber.txt #doc doc/jingle.txt %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/gtalk.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/jabber.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/jingle.conf %{_libdir}/asterisk/modules/chan_gtalk.so %{_libdir}/asterisk/modules/chan_jingle.so %{_libdir}/asterisk/modules/res_jabber.so %files jack %defattr(-,root,root,-) %{_libdir}/asterisk/modules/app_jack.so %files lua %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/extensions.lua %{_libdir}/asterisk/modules/pbx_lua.so %files ldap %defattr(-,root,root,-) #doc doc/ldap.txt %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_ldap.conf %{_libdir}/asterisk/modules/res_config_ldap.so %if 0%{?rhel} <= 5 || 0%{?fedora} %files ldap-389 %defattr(-,root,root,-) %{_sysconfdir}/dirsrv/schema/99asterisk.ldif %endif %files minivm %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/extensions_minivm.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/minivm.conf %{_libdir}/asterisk/modules/app_minivm.so %files misdn %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/misdn.conf #%{_libdir}/asterisk/modules/chan_misdn.so %files mobile %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/chan_mobile.conf %{_libdir}/asterisk/modules/chan_mobile.so %files mysql %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/app_mysql.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cdr_mysql.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_config_mysql.conf %doc contrib/realtime/mysql/*.sql %{_libdir}/asterisk/modules/app_mysql.so %{_libdir}/asterisk/modules/cdr_mysql.so %{_libdir}/asterisk/modules/res_config_mysql.so %files odbc %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cdr_adaptive_odbc.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cdr_odbc.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cel_odbc.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/func_odbc.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_odbc.conf %{_libdir}/asterisk/modules/cdr_adaptive_odbc.so %{_libdir}/asterisk/modules/cdr_odbc.so %{_libdir}/asterisk/modules/cel_odbc.so %{_libdir}/asterisk/modules/func_odbc.so %{_libdir}/asterisk/modules/res_config_odbc.so %{_libdir}/asterisk/modules/res_odbc.so %files ooh323 %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/chan_ooh323.conf %{_libdir}/asterisk/modules/chan_ooh323.so %files oss %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/oss.conf %{_libdir}/asterisk/modules/chan_oss.so %files portaudio %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/console.conf %{_libdir}/asterisk/modules/chan_console.so %files postgresql %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cdr_pgsql.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cel_pgsql.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_pgsql.conf %doc contrib/realtime/postgresql/*.sql %{_libdir}/asterisk/modules/cdr_pgsql.so %{_libdir}/asterisk/modules/cel_pgsql.so %{_libdir}/asterisk/modules/res_config_pgsql.so %files radius %defattr(-,root,root,-) %{_libdir}/asterisk/modules/cdr_radius.so %{_libdir}/asterisk/modules/cel_radius.so %files skinny %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/skinny.conf %{_libdir}/asterisk/modules/chan_skinny.so %files snmp %defattr(-,root,root,-) #doc doc/asterisk-mib.txt #doc doc/digium-mib.txt #doc doc/snmp.txt %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_snmp.conf #%{_datadir}/snmp/mibs/ASTERISK-MIB.txt #%{_datadir}/snmp/mibs/DIGIUM-MIB.txt %{_libdir}/asterisk/modules/res_snmp.so %files sqlite %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cdr_sqlite3_custom.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cel_sqlite3_custom.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_config_sqlite.conf %{_libdir}/asterisk/modules/cdr_sqlite3_custom.so %{_libdir}/asterisk/modules/cel_sqlite3_custom.so %files tds %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cdr_tds.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/cel_tds.conf %{_libdir}/asterisk/modules/cdr_tds.so %{_libdir}/asterisk/modules/cel_tds.so %files unistim %defattr(-,root,root,-) #doc doc/unistim.txt %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/unistim.conf %{_libdir}/asterisk/modules/chan_unistim.so %files voicemail %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/voicemail.conf %{_libdir}/asterisk/modules/func_vmcount.so %if 0%{?fedora} > 0 %files voicemail-imap %defattr(-,root,root,) %{_libdir}/asterisk/modules/app_directory_imap.so %{_libdir}/asterisk/modules/app_voicemail_imap.so %endif %files voicemail-odbc %defattr(-,root,root,-) #doc doc/voicemail_odbc_postgresql.txt %{_libdir}/asterisk/modules/app_directory_odbc.so %{_libdir}/asterisk/modules/app_voicemail_odbc.so %files voicemail-plain %defattr(-,root,root,-) %{_libdir}/asterisk/modules/app_directory_plain.so %{_libdir}/asterisk/modules/app_voicemail_plain.so %changelog * Thu May 3 2012 Jeffrey Ollie - 1.8.12.0-1: - The Asterisk Development Team has announced the release of Asterisk 1.8.12.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - 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 - (Closes issue ASTERISK-19493. Reported by lvl) - - * --- Fix Dial m and r options and forked calls generating warnings - for voice frames. - (Closes issue ASTERISK-16901. Reported by Chris Gentle) - - * --- Remove ISDN hold restriction for non-bridged calls. - (Closes issue ASTERISK-19388. Reported by Birger Harzenetter) - - * --- Fix copying of CDR(accountcode) to local channels. - (Closes issue ASTERISK-19384. Reported by jamicque) - - * --- Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors - (Closes issue ASTERISK-19303. Reported by Jon Tsiros) - - * --- Eliminate double close of file descriptor in manager.c - (Closes issue ASTERISK-18453. Reported by Jaco Kroon) - - 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 * Tue Apr 24 2012 Jeffrey Ollie - 1.8.11.1-1: - 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. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - 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 the - 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 * Fri Mar 30 2012 Russell Bryant - 1.8.11.0-1 - Update to 1.8.11.0 * Sat Mar 17 2012 Russell Bryant - 1.8.10.1-1 - Update to 1.8.10.1 from upstream. - Fix remote stack overflow in app_milliwatt. - Fix remote stack overflow, including possible code injection, in HTTP digest authentication handling. - Diable build of SRTP on ppc64, as it doesn't build right now. - Resolves: rhbz#804045, rhbz#804038, rhbz#804042 * Thu Nov 17 2011 Jeffrey C. Ollie - 1.8.8.0-0.4.rc4 - The Asterisk Development Team has announced the fourth release candidate of - Asterisk 1.8.8.0. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.8.0-rc4 resolves a particular issue with BLF - subscriptions. A change in Asterisk 1.8.8.0-rc3 had the potential to cause a - segfault, and this release candidate was created to resolve that. - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0-rc4 * Thu Nov 10 2011 Jeffrey C. Ollie - 1.8.8.0-0.3.rc3 - The Asterisk Development Team has announced the third release candidate of - Asterisk 1.8.8.0. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.8.0-rc3 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 candidate: - - * Prevent BLF subscriptions from causing deadlocks. - (Closes issue ASTERISK-18663) - Review: https://reviewboard.asterisk.org/r/1563/ - - * Fix deadlock if peer is destroyed while sending MWI notice. - (Closes issue ASTERISK-18747) - Reported by: Gregory Hinton Nietsky - - * Fix issue with setting defaultenabled on categories that are already enabled - by default. - (Closes issue ASTERISK-18738) - Reported by: Paul Belanger - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0-rc3 * Tue Nov 8 2011 Jeffrey C. Ollie - 1.8.8.0-0.2.rc2 - The Asterisk Development Team has announced the second release candidate of - Asterisk 1.8.8.0. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.8.0-rc2 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 candidate: - - * --- 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 --- - (Closes issue ASTERISK-18101. Reported by Paul Rolfe, patched by Gregory Nietsky) - (Closes issue ASTERISK-18487. Reported by Jason Legault, patched by Gregory - Nietsky) - - * --- Fix regression in configure script for libpri capability checks --- - (Closes issue ASTERISK-18687. Reported by norbert, patched by Richard Mudgett) - - * --- Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more places --- - (Closes issue ASTERISK-18610. Reported by Kristijan_Vrban, patched by Terry - Wilson, and again by Kristijan_Vrban) - - * --- Fix issue with removing peers by IP --- - (Closes issue ASTERISK-18696. Reported by rsw686, patched by Terry Wilson) - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0-rc2 * Tue Nov 8 2011 Jeffrey C. Ollie - 1.8.8.0-0.1.rc1 - The Asterisk Development Team announces the first release candidate of - Asterisk 1.8.8.0. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.8.0-rc1 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 candidate: - - * 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. - (Closes ASTERISK-17288. Reported by: Mikael Carlsson Tested by: Matthew - Jordan Review: https://reviewboard.asterisk.org/r/1416/) - - * 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. - (Closes issue ASTERISK-18090. Patched by Kinsey Moore) - - * Fix bad RTP media bridges in directmedia calls on peers separated by multiple - Asterisk nodes. - (Closes issue ASTERISK-18340. Reported by: Thomas Arimont. Closes issue - ASTERISK-17725. Reported by: kwk. Tested by: twilson, jrose) - - * Fix crashes in ast_rtcp_write() - (Closes issue ASTERISK-18570) - Related issues that look like they are the same problem: - (Issue ASTERISK-17560, ASTERISK-15406, ASTERISK-15257, ASTERISK-13334, - ASTERISK-9977, ASTERISK-9716) - Review: https://reviewboard.asterisk.org/r/1444/ - Patched by: Russell Bryant - - * 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. - (Closes ASTERISK-16981. Reported by: Mary Ciuciu, Byron Clark, Brad House, - Karsten Wemheuer, KevinH Tested by: Matt Jordan - Review: https://reviewboard.asterisk.org/r/1443) - - * Prevent segfault if call arrives before Asterisk is fully booted. - (Patched by alecdavis. https://reviewboard.asterisk.org/r/1407/) - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.8.0-rc1 * Mon Oct 17 2011 Jeffrey C. Ollie - 1.8.7.1-1 - The Asterisk Development Team has announced a security release for Asterisk 1.8. - The available security release is released as version 1.8.7.1. - - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.8.7.1 resolves an issue with SIP URI parsing which can - lead to a remotely exploitable crash: - - Remote Crash Vulnerability in SIP channel driver (AST-2011-012) - - The issue and resolution is described in the AST-2011-012 security - advisory. - - For more information about the details of this vulnerability, please read the - security advisory AST-2011-012, 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-1.8.7.1 * Mon Oct 3 2011 Jeffrey C. Ollie - 1.8.7.0-1 - The Asterisk Development Team announces the release of Asterisk 1.8.7.0. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - 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. Reported by: David Cabrejos, Remi Quezada, - Philippe Lindheimer, David Woolley, Mat Murdock. Patched by: rmudgett) - - * 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. Reported, patched by: Michael L. Young. Tested - by Jonathan Rose) - - * 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. Reported by Sean Darcy. Patched by Richard - Mudgett) - - * Remove unnecessary libpri dependency checks in the configure script. - (Closes issue ASTERISK-18535. Reported by Michael Keuter. Patched by Richard - Mudgett) - - * 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 * Tue Sep 20 2011 Jeffrey C. Ollie - 1.8.6.0-4 - Add additional patch for res_pktccops. * Tue Sep 20 2011 Jeffrey C. Ollie - 1.8.6.0-3 - Add patch to fix compatibility with 389 directory server. * Tue Sep 20 2011 Jeffrey C. Ollie - 1.8.6.0-2 - Add patches to fix many bug reports from bugzilla. * Tue Sep 20 2011 Jeffrey C. Ollie - 1.8.6.0-1 - The Asterisk Development Team announces the release of Asterisk 1.8.6.0. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - 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. Reported by David Cunningham. Patched by Igor - Goncharovsky) - - * Resolve a potential crash in chan_sip when utilizing auth= and performing a - 'sip reload' from the console. - (Closes issue ASTERISK-17939. Reported by wdoekes. Patched by Richard Mudgett) - - * 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. Reported by marcelloceschia. Patched by Tilghman - Lesher) - - * Resolve issue where 403 Forbidden would always be sent maximum number of times - regardless to receipt of ACK. - (Patched by Richard Mudgett) - - * 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. - (Patched by Kinsey Moore) - - * Fix New Zealand indications profile based on - http://www.telepermit.co.nz/TNA102.pdf - (Closes issue ASTERISK-16263. Reported, Patched by richardf) - - * Segfault in shell_helper in func_shell.c - (Closes issue ASTERISK-18109. Reported by Michael Myles, patched by Richard - Mudgett) - - 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 * Tue Aug 23 2011 Jeffrey C. Ollie - 1.8.6.0-0.2.rc2 - The Asterisk Development Team has announced the second release candidate of - Asterisk 1.8.6.0. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.6.0-rc2 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 candidate: - - * --- Segfault in shell_helper in func_shell.c --- - (Closes issue ASTERISK-18109. - Reported by Michael Myles, patched by Richard Mudgett) - - * --- Re-add support for spaces in pathnames --- - (Closes issue ASTERISK-18290. - Reported by Paul Belanger, patched by Tilghman Lesher) - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.6.0-rc2 * Thu Aug 11 2011 Jeffrey C. Ollie - 1.8.6.0-0.1.rc1 - The Asterisk Development Team announces the first release candidate of - Asterisk 1.8.6.0. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.6.0-rc1 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 candidate: - - * Fix an issue with Music on Hold classes losing files in playlist when realtime - is used. - (Closes issue ASTERISK-17875. Reported by David Cunningham. Patched by Igor - Goncharovsky) - - * Resolve a potential crash in chan_sip when utilizing auth= and performing a - 'sip reload' from the console. - (Closes issue ASTERISK-17939. Reported by wdoekes. Patched by Richard Mudgett) - - * 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. Reported by marcelloceschia. Patched by Tilghman - Lesher) - - * Resolve issue where 403 Forbidden would always be sent maximum number of times - regardless to receipt of ACK. - (Patched by Richard Mudgett) - - * Updated chan_gtalk to work with changes made by Google. - (Closes issue ASTERISK-18804. Patched by Terry Wilson) - - * 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. - (Patched by Kinsey Moore) - - * Fix New Zealand indications profile based on - http://www.telepermit.co.nz/TNA102.pdf - (Closes issue ASTERISK-16263. Reported, Patched by richardf) - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.6.0-rc1 * Thu Jul 21 2011 Petr Sabata - 1.8.5.0-1.2 - Perl mass rebuild * Wed Jul 20 2011 Petr Sabata - 1.8.5.0-1.1 - Perl mass rebuild * Mon Jul 11 2011 Jeffrey C. Ollie - 1.8.5.0-1 - The Asterisk Development Team announces the release of Asterisk 1.8.5.0. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - 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 - (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, - cmaj) - - * Fixes thread blocking issue in the sip TCP/TLS implementation. - (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, - rossbeer, kowalma, Freddi_Fonet) - - * Be more tolerant of what URI we accept for call completion PUBLISH requests. - (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson) - - * Fix a nasty chanspy bug which was causing a channel leak every time a spied on - channel made a call. - (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose) - - * This patch fixes a bug with MeetMe behavior where the 'P' option for always - prompting for a pin is ignored for the first caller. - (Closes issue #18070. Reported by mav3rick. Patched by bbryant) - - * 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. - (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett) - - * 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. - (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard - Mudgett) - - * 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. - (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett) - - 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 * Thu Jul 7 2011 Jeffrey C. Ollie - 1.8.5-0.2 - Rebuild for net-snmp 5.7 * Fri Jul 1 2011 Jeffrey C. Ollie - 1.8.5-0.1.rc1 - Fix systemd dependencies in EL6 and F15 * Fri Jul 1 2011 Jeffrey C. Ollie - 1.8.4.4-3 - Bump release * Fri Jul 1 2011 Jeffrey C. Ollie - 1.8.4.4-2 - Fix systemd dependencies in EL6 and F15 * Thu Jun 30 2011 Jeffrey C. Ollie - 1.8.5-0.1.rc1 - The Asterisk Development Team has announced the first release candidate of - Asterisk 1.8.5. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.5-rc1 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 candidate: - - * Fix Deadlock with attended transfer of SIP call - (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81, - cmaj) - - * Fixes thread blocking issue in the sip TCP/TLS implementation. - (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois, - rossbeer, kowalma, Freddi_Fonet) - - * Be more tolerant of what URI we accept for call completion PUBLISH requests. - (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson) - - * Fix a nasty chanspy bug which was causing a channel leak every time a spied on - channel made a call. - (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose) - - * This patch fixes a bug with MeetMe behavior where the 'P' option for always - prompting for a pin is ignored for the first caller. - (Closes issue #18070. Reported by mav3rick. Patched by bbryant) - - * 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. - (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett) - - * 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. - (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard - Mudgett) - - * 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. - (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett) - - * Fix timerfd locking issue. - (Closes ASTERISK-17867, ASTERISK-17415. Patched by kobaz) - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5-rc1 * Thu Jun 30 2011 Jeffrey C. Ollie - 1.8.4.4-2 - Fedora Directory Server -> 389 Directory Server * Wed Jun 29 2011 Jeffrey C. Ollie - 1.8.4.4-1 - The Asterisk Development Team has announced the release of Asterisk - versions 1.4.41.2, 1.6.2.18.2, and 1.8.4.4, which are security - releases. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 resolves the - following issue: - - AST-2011-011: 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. - - For more information about the details of this vulnerability, please - read the security advisory AST-2011-011, which was 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-1.4.41.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.18.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.4 - - Security advisory AST-2011-011 is available at: - - http://downloads.asterisk.org/pub/security/AST-2011-011.pdf * Mon Jun 27 2011 Jeffrey C. Ollie - 1.8.4.3-3 - Don't forget stereorize * Mon Jun 27 2011 Jeffrey C. Ollie - 1.8.4.3-2 - Move /var/run/asterisk to /run/asterisk - Add comments to systemd service file on how to mimic safe_asterisk functionality - Build more of the optional binaries - Install the tmpfiles.d configuration on Fedora 15 * Fri Jun 24 2011 Jeffrey C. Ollie - 1.8.4.3-1 - The Asterisk Development Team has announced the release of Asterisk versions - 1.4.41.1, 1.6.2.18.1, and 1.8.4.3, which are security releases. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.4.41.1, 1.6.2.18, and 1.8.4.3 resolves several issues - as outlined below: - - * AST-2011-008: If a remote user sends a SIP packet containing a null, - Asterisk assumes available data extends past the null to the - end of the packet when the buffer is actually truncated when - copied. This causes SIP header parsing to modify data past - the end of the buffer altering unrelated memory structures. - This vulnerability does not affect TCP/TLS connections. - -- Resolved in 1.6.2.18.1 and 1.8.4.3 - - * AST-2011-009: A remote user sending a SIP packet containing a Contact header - with a missing left angle bracket (<) causes Asterisk to - access a null pointer. - -- Resolved in 1.8.4.3 - - * AST-2011-010: A memory address was inadvertently transmitted over the - network via IAX2 via an option control frame and the remote party would try - to access it. - -- Resolved in 1.4.41.1, 1.6.2.18.1, and 1.8.4.3 - - The issues and resolutions are described in the AST-2011-008, AST-2011-009, and - AST-2011-010 security advisories. - - For more information about the details of these vulnerabilities, please read - the security advisories AST-2011-008, AST-2011-009, and AST-2011-010, 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-1.4.41.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.18.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.3 - - Security advisories AST-2011-008, AST-2011-009, and AST-2011-010 are available - at: - - http://downloads.asterisk.org/pub/security/AST-2011-008.pdf - http://downloads.asterisk.org/pub/security/AST-2011-009.pdf - http://downloads.asterisk.org/pub/security/AST-2011-010.pdf * Tue Jun 21 2011 Jeffrey C. Ollie - 1.8.4.2-2 - Convert to systemd * Fri Jun 17 2011 Marcela Mašláňová - 1.8.4.2-1.2 - Perl mass rebuild * Fri Jun 10 2011 Marcela Mašláňová - 1.8.4.2-1.1 - Perl 5.14 mass rebuild * Fri Jun 3 2011 Jeffrey C. Ollie - 1.8.4.2-1: - - The Asterisk Development Team has announced the release of Asterisk - version 1.8.4.2, which is a security release for Asterisk 1.8. - - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - 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 - - The Asterisk Development Team has announced the release of Asterisk 1.8.4.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - 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) - (Closes issue #18951. Reported by jmls. Patched by wdoekes) - - * 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. - (Closes issue #18951. Patched by mnicholson) - - * Resolve potential crash when using SIP TLS support. - (Closes issue #19192. Reported by stknob. Patched by Chainsaw. Tested by - vois, Chainsaw) - - * Improve reliability when using SIP TLS. - (Closes issue #19182. Reported by st. Patched by mnicholson) - - - 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 - The Asterisk Development Team has announced the release of Asterisk 1.8.4. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - 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. - (Closes issue #19095, #19138. Reported, patched by tzafrir. Tested by russell - and chazzam. - - * Resolve crash in ast_mutex_init() - (Patched by twilson) - - * Resolution of several DTMF based attended transfer issues. - (Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo, - shihchuan, grecco. Patched by rmudgett) - - NOTE: Be sure to read the ChangeLog for more information about these changes. - - * Resolve deadlocks related to device states in chan_sip - (Closes issue #18310. Reported, patched by one47. Patched by jpeeler) - - * Resolve an issue with the Asterisk manager interface leaking memory when - disabled. - (Reported internally by kmorgan. Patched by russellb) - - * Support greetingsfolder as documented in voicemail.conf.sample. - (Closes issue #17870. Reported by edhorton. Patched by seanbright) - - * Fix channel redirect out of MeetMe() and other issues with channel softhangup - (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb. - Patched by russellb) - - * Fix voicemail sequencing for file based storage. - (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by - jpeeler) - - * 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+. - (Patched by twilson) - - * Fix issues with verbose messages not being output to the console. - (Closes issue #18580. Reported by pabelanger. Patched by qwell) - - * Fix Deadlock with attended transfer of SIP call - (Closes issue #18837. Reported, patched by alecdavis. Tested by - alecdavid, Irontec, ZX81, cmaj) - - 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 * Thu Apr 21 2011 Jeffrey C. Ollie - 1.8.3.3-1 - 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. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - 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-1.4.40.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.25 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.3 - 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 * Wed Mar 23 2011 Jeffrey C. Ollie - 1.8.3.2-2 - Bump release and rebuild for mysql 5.5.10 soname change. * Thu Mar 17 2011 Jeffrey C. Ollie - 1.8.3.2-1 - 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 1.6.1.24, 1.6.2.17.2, and 1.8.3.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - ** 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-1.6.1.24 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.2 - 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 * Thu Mar 17 2011 Jeffrey C. Ollie - 1.8.3.1-1 - 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. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - 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-1.6.1.23 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.1 - 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 * Mon Feb 28 2011 - 1.8.3-1 - The Asterisk Development Team has announced the release of Asterisk 1.8.3. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - 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() - (Closes issue #18091. Reported by bunny. Patched by tilghman) - - * Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. - (Closes issue #18464. Reported, patched by IgorG) - - * Reworking parsing of mwi => lines to resolve a segfault. Also add a set of - unit tests for the function that does the parsing. - (Closes issue #18350. Reported by gbour. Patched by Marquis) - - * When using cdr_pgsql the billsec field was not populated correctly on - unanswered calls. - (Closes issue #18406. Reported by joscas. Patched by tilghman) - - * Resolve memory leak in iCalendar and Exchange calendaring modules. - (Closes issue #18521. Reported, patched by pitel. Tested by cervajs) - - * 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. - (Patched by tilghman) - - * Resolve issue where no Music On Hold may be triggered when using - res_timing_dahdi. - (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested - by francesco_r, rfrantik, one47) - - * Resolve a memory leak when the Asterisk Manager Interface is disabled. - (Reported internally by kmorgan. Patched by russellb) - - * Reimplemented fax session reservation to reverse the ABI breakage introduced - in r297486. - (Reported internally. Patched by mnicholson) - - * Fix regression that changed behavior of queues when ringing a queue member. - (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.) - - * Resolve deadlock involving REFER. - (Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.) - - 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 * Wed Feb 16 2011 - 1.8.3-0.7.rc3 - - The Asterisk Development Team has announced the third release candidate of - Asterisk 1.8.3. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.3-rc3 resolves the following issues in addition to - those included in 1.8.3-rc1 and 1.8.3-rc2: - - * Fix regression that changed behavior of queues when ringing a queue member. - (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.) - - * Resolve deadlock involving REFER. - (Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.) - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3-rc3 * Fri Feb 11 2011 Jeffrey C. Ollie - 1.8.3-0.6.rc2 - Bump release to build for F15 * Wed Feb 9 2011 Jeffrey C. Ollie - 1.8.3-0.5.rc2 - Remove isa macros * Wed Feb 9 2011 Jeffrey C. Ollie - 1.8.3-0.4.rc2 - Make library dependencies architecture specific * Mon Feb 07 2011 Fedora Release Engineering - 1.8.3-0.3.rc2 - Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild * Wed Jan 26 2011 Jeffrey C. Ollie - 1.8.3-0.2.rc2 The Asterisk Development Team has announced the second release candidate of Asterisk 1.8.3. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.3-rc2 resolves the following issues in addition to those included in 1.8.3-rc1: * Resolve issue where no Music On Hold may be triggered when using res_timing_dahdi. (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested by francesco_r, rfrantik, one47) * Resolve a memory leak when the Asterisk Manager Interface is disabled. (Reported internally by kmorgan. Patched by russellb) * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported internally. Patched by mnicholson) For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3-rc2 * Wed Jan 26 2011 Jeffrey C. Ollie - 1.8.3-0.1.rc1 - - The Asterisk Development Team has announced the first release candidate of - Asterisk 1.8.3. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 1.8.3-rc1 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 candidate: - - * Resolve duplicated data in the AstDB when using DIALGROUP() - (Closes issue #18091. Reported by bunny. Patched by tilghman) - - * Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. - (Closes issue #18464. Reported, patched by IgorG) - - * Reworking parsing of mwi => lines to resolve a segfault. Also add a set of - unit tests for the function that does the parsing. - (Closes issue #18350. Reported by gbour. Patched by Marquis) - - * When using cdr_pgsql the billsec field was not populated correctly on - unanswered calls. - (Closes issue #18406. Reported by joscas. Patched by tilghman) - - * Resolve memory leak in iCalendar and Exchange calendaring modules. - (Closes issue #18521. Reported, patched by pitel. Tested by cervajs) - - * 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. - (Patched by tilghman) * Wed Jan 26 2011 Jeffrey C. Ollie - 1.8.2.3-1 - - The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - 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 * Mon Jan 24 2011 Jeffrey C. Ollie - 1.8.2.2-2 - Build with SRTP support * Mon Jan 24 2011 Jeffrey C. Ollie - 1.8.2.2-1 - - The Asterisk Development Team has announced a release for the security issue - described in AST-2011-001. - - Due to a failed merge, Asterisk 1.8.2.1 which should have included the security - fix did not. Asterisk 1.8.2.2 contains the the changes which should have been - included in Asterisk 1.8.2.1. - - This releases is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2, - 1.8.1.2, and 1.8.2.2 resolve an issue when forming an outgoing SIP request while - in pedantic mode, which can cause a stack buffer to be made to overflow if - supplied with carefully crafted caller ID information. The issue and resolution - are described in the AST-2011-001 security advisory. - - For more information about the details of this vulnerability, please read the - security advisory AST-2011-001, 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-1.8.2.2 - - Security advisory AST-2011-001 is available at: - - http://downloads.asterisk.org/pub/security/AST-2011-001.pdf * Mon Jan 24 2011 Jeffrey C. Ollie - 1.8.2.1-1 - - The Asterisk Development Team has announced security releases for the following - versions of Asterisk: - - * 1.4.38.1 - * 1.4.39.1 - * 1.6.1.21 - * 1.6.2.15.1 - * 1.6.2.16.1 - * 1.8.1.2 - * 1.8.2.1 - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2, - 1.8.1.2, and 1.8.2.1 resolve an issue when forming an outgoing SIP request while - in pedantic mode, which can cause a stack buffer to be made to overflow if - supplied with carefully crafted caller ID information. The issue and resolution - are described in the AST-2011-001 security advisory. - - For more information about the details of this vulnerability, please read the - security advisory AST-2011-001, which was 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-1.4.38.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.39.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.21 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.15.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.16.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.1.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.1 - - Security advisory AST-2011-001 is available at: - - http://downloads.asterisk.org/pub/security/AST-2011-001.pdf * Mon Jan 24 2011 Jeffrey C. Ollie - 1.8.2-1 - - The Asterisk Development Team has announced the release of Asterisk 1.8.2. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - 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 * Mon Jan 24 2011 Jeffrey C. Ollie - 1.8.1.1-1 - - The Asterisk Development Team has announced the release of Asterisk 1.8.1.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - 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 * Mon Jan 24 2011 Jeffrey C. Ollie - 1.8.1-1 - - The Asterisk Development Team has announced the release of Asterisk 1.8.1. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - 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 * Tue Jan 18 2011 Dennis Gilmore - 1.8.0-6 - dont package up the ices bits on el the client doesnt exist for us * Tue Jan 18 2011 Dennis Gilmore - 1.8.0-5 - dont build the 389 directory server package its not available on rhel6 * Fri Dec 10 2010 Dennis Gilmore - 1.8.0-4 - dont always build AIS modules we dont have the BuildRequires on epel * Fri Oct 29 2010 Jeffrey C. Ollie - 1.8.0-3 - Rebuild for new net-snmp. * Tue Oct 26 2010 Jeffrey C. Ollie - 1.8.0-2 - Always build AIS modules * Thu Oct 21 2010 Jeffrey C. Ollie - 1.8.0-1 - The Asterisk Development Team is proud to announce the release of Asterisk - 1.8.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - Asterisk 1.8 is the next major release series of Asterisk. It will be a Long - Term Support (LTS) release, similar to Asterisk 1.4. For more information about - support time lines for Asterisk releases, see the Asterisk versions page. - - http://www.asterisk.org/asterisk-versions - - 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 - - Thank you for your continued support of Asterisk! * Mon Oct 18 2010 Jeffrey C. Ollie - 1.8.0-0.8.rc5: - - The release of Asterisk 1.8.0-rc5 was triggered by some last minute platform - compatibility IPv6 changes. In addition, the availability of the English sound - prompts with Australian accents has been added. - - 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-rc5 - - This release candidate contains fixes since the last release candidate as - reported by the community. A sampling of the changes in this release candidate - include: - - * Additional fixups in chan_gtalk that allow outbound calls to both Google - Talk and Google Voice recipients. Adds new chan_gtalk enhancements externip - and stunaddr. - (Closes issue #13971. Patched by dvossel) - - * Resolve manager crash issue. - (Closes issue #17994. Reported by vrban. Patchd by dvossel) - - * Documentation updates for sample configuration files. - (Closes issues #18107, #18101. Reported, patched by lathama, lmadsen) - - * Resolve issue where faxdetect would only detect the first fax call in - chan_dahdi. - (Closes issue #18116. Reported by seandarcy. Patched by rmudgett) - - * Resolve issue where a channel that is setup and torn down *very* quickly may - not have the right call disposition or ${DIALSTATUS}. - (Closes issue #16946. Reported by davidw. Review - https://reviewboard.asterisk.org/r/740/) - - * Set TCLASS field of IPv6 header when SIP QoS options are set. - (Closes issue #18099. Reported by jamesnet. Patched by dvossel) - - * Resolve issue where Asterisk could crash on shutdown when using SRTP. - (Closes issue #18085. Reported by st. Patched by twilson) - - * Fix issue where peers host port would be lost on a SIP reload. - (Closes issue #18135. Reported, tested by lmadsen. Patched by dvossel) - - 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-rc4 * Fri Oct 8 2010 Jeffrey C. Ollie - 1.8.0-0.7.rc3 - This release candidate contains fixes since the release candidate as reported by - the community. A sampling of the changes in this release candidate include: - - * Still build chan_sip even if res_crypto cannot be built (use, but not depend) - (Reported by a user on the mailing list. Patched by tilghman) - - * Get notifications for call files only when a file is closed, not when created - (Closes issue #17924. Reported by mkeuter. Patched by abeldeck) - - * Fixes to chan_gtalk to allow outbound DTMF support to work correctly. Gtalk - expects the DTMF to arrive on the RTP stream and not via jingle DTMF - signalling. - (Patched by dvossel. Tested by malcolmd) - - * Fixes to allow chan_gtalk to communicate with the Gmail web client. - (Patched by phsultan and dvossel) - - * Fix to GET DATA to allow audio to be streamed via an AGI. - (Closes issue #18001. Reported by jamicque. Patched by tilghman) - - * Resolve dnsmgr memory corruption in chan_iax2. - (Closes issue #17902. Reported by afried. Patched by russell, dvossel) - - 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=checkout - - 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-rc3 * Wed Oct 6 2010 Jeffrey C. Ollie - 1.8.0-0.6.rc2 - This release candidate contains fixes since the last beta release as reported by - the community. A sampling of the changes in this release candidate include: - - * Add slin16 support for format_wav (new wav16 file extension) - (Closes issue #15029. Reported, patched by andrew. Tested by Qwell) - - * Fixes a bug in manager.c where the default configuration values weren't reset - when the manager configuration was reloaded. - (Closes issue #17917. Reported by lmadsen. Patched by bbryant) - - * Various fixes for the calendar modules. - (Patched by Jan Kalab. - Reviewboard: https://reviewboard.asterisk.org/r/880/ - Closes issue #17877. Review: https://reviewboard.asterisk.org/r/916/ - Closes issue #17776. Review: https://reviewboard.asterisk.org/r/921/) - - * Add CHANNEL(checkhangup) to check whether a channel is in the process of - being hung up. - (Closes issue #17652. Reported, patched by kobaz) - - * Fix a bug with MeetMe where after announcing the amount of time left in a - conference, if music on hold was playing, it doesn't restart. - (Closes issue #17408, Reported, patched by sysreq) - - * Fix interoperability problems with session timer behavior in Asterisk. - (Closes issue #17005. Reported by alexcarey. Patched by dvossel) - - * Rate limit calls to fsync() to 1 per second after astdb updates. Astdb was - determined to be one of the most significant bottlenecks in SIP registration - processing. This patch improved the speed of an astdb load test by 50000% - (yes, Fifty-Thousand Percent). On this particular load test setup, this - doubled the number of SIP registrations the server could handle. - (Review: https://reviewboard.asterisk.org/r/825/) - - * Don't clear the username from a realtime database when a registration - expires. Non-realtime chan_sip does not clear the username from memory when a - registration expiries so realtime probably shouldn't either. - (Closes issue #17551. Reported, patched by: ricardolandim. Patched by - mnicholson) - - * Don't hang up a call on an SRTP unprotect failure. Also make it more obvious - when there is an issue en/decrypting. - (Closes issue #17563. Reported by Alexcr. Patched by sfritsch. Tested by - twilson) - - * Many more issues. This is a significant upgrade over Asterisk 1.8.0 beta 5! - - 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=checkout - - 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-rc2 * Thu Sep 9 2010 Jeffrey C. Ollie - 1.8.0-0.5.beta5 - This release contains fixes since the last beta release as reported by the - community. A sampling of the changes in this release include: - - * Fix issue where TOS is no longer set on RTP packets. - (Closes issue #17890. Reported, patched by elguero) - - * Change pedantic default value in chan_sip from 'no' to 'yes' - - * Asterisk now dynamically builds the "Supported" header depending on what is - enabled/disabled in sip.conf. Session timers used to always be advertised as - being supported even when they were disabled in the configuration. - (Related to issue #17005. Patched by dvossel) - - * Convert MOH to use generic timers. - (Closes issue #17726. Reported by lmadsen. Patched by tilghman) - - * Fix SRTP for changing SSRC and multiple a=crypto SDP lines. Adding code to - Asterisk that changed the SSRC during bridges and masquerades broke SRTP - functionality. Also broken was handling the situation where an incoming - INVITE had more than one crypto offer. - (Closes issue #17563. Reported by Alexcr. Patched by twilson) - - Asterisk 1.8 contains many new features over previous releases of Asterisk. - A short list of included features includes: - - * Secure RTP - * IPv6 Support in the SIP Channel - * 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=checkout - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta5 * Tue Aug 24 2010 Jeffrey C. Ollie - 1.8.0-0.4.beta4 - This release contains fixes since the last beta release as reported by the - community. A sampling of the changes in this release include: - - * Fix parsing of IPv6 address literals in outboundproxy - (Closes issue #17757. Reported by oej. Patched by sperreault) - - * Change the default value for alwaysauthreject in sip.conf to "yes". - (Closes issue #17756. Reported by oej) - - * Remove current STUN support from chan_sip.c. This change removes the current - broken/useless STUN support from chan_sip. - (Closes issue #17622. Reported by philipp2. - Review: https://reviewboard.asterisk.org/r/855/) - - * PRI CCSS may use a stale dial string for the recall dial string. If an - outgoing call negotiates a different B channel than initially requested, the - saved original dial string was not transferred to the new B channel. CCSS - uses that dial string to generate the recall dial string. - (Patched by rmudgett) - - * Split _all_ arguments before parsing them. This fixes multicast RTP paging - using linksys mode. - (Patched by russellb) - - * Expand cel_custom.conf.sample. Include the usage of CSV_QUOTE() to ensure - data has valid CSV formatting. Also list the special CEL variables that are - available for use in the mapping. There are also several other CEL fixes in - this release. - (Patched by russellb) - - Asterisk 1.8 contains many new features over previous releases of Asterisk. - A short list of included features includes: - - * Secure RTP - * IPv6 Support in the SIP Channel - * 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=checkout - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta4 * Wed Aug 11 2010 Jeffrey C. Ollie - 1.8.0-0.3.beta3 - - This release contains fixes since the last beta release as reported by the - community. A sampling of the changes in this release include: - - * Fix a regression where HTTP would always be enabled regardless of setting. - (Closes issue #17708. Reported, patched by pabelanger) - - * ACL errors displayed on screen when using dynamic_exclude_static in sip.conf - (Closes issue #17717. Reported by Dennis DeDonatis. Patched by mmichelson) - - * Support "channels" in addition to "channel" in chan_dahdi.conf. - (https://reviewboard.asterisk.org/r/804) - - * Fix parsing error in sip_sipredirect(). The code was written in a way that - did a bad job of parsing the port out of a URI. Specifically, it would do - badly when dealing with an IPv6 address. - (Closes issue #17661. Reported by oej. Patched by mmichelson) - - * Fix inband DTMF detection on outgoing ISDN calls. - (Patched by russellb and rmudgett) - - * Fixes issue with translator frame not getting freed. This issue prevented - g729 licenses from being freed up. - (Closes issue #17630. Reported by manvirr. Patched by dvossel) - - * Fixed IPv6-related SIP parsing bugs and updated documention. - (Reported by oej. Patched by sperreault) - - * Add new, self-contained feature FIELDNUM(). Returns a 1-based index into a - list of a specified item. Matches up with FIELDQTY() and CUT(). - (Closes #17713. Reported, patched by gareth. Tested by tilghman) - - Asterisk 1.8 contains many new features over previous releases of Asterisk. - A short list of included features includes: - - * Secure RTP - * IPv6 Support - * 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=checkout - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta3 * Mon Aug 2 2010 Jeffrey C. Ollie - 1.8.0-0.2.beta2 - Rebuild against libpri 1.4.12 * Mon Aug 2 2010 Jeffrey C. Ollie - 1.8.0-0.1.beta2 - Update to 1.8.0-beta2 - Disable building chan_misdn until compilation errors are figured out (https://issues.asterisk.org/view.php?id=14333) - Start stripping tarballs again because Digium added MP3 code :( * Sat Jul 31 2010 Jeffrey C. Ollie - 1.6.2.10-1 - - The following are a few of the issues resolved by community developers: - - * Allow users to specify a port for DUNDI peers. - (Closes issue #17056. Reported, patched by klaus3000) - - * Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is - set. - (Closes issue #16815. Reported, patched by rain) - - * If there is realtime configuration, it does not get re-read on reload unless - the config file also changes. - (Closes issue #16982. Reported, patched by dmitri) - - * Send AgentComplete manager event for attended transfers. - (Closes issue #16819. Reported, patched by elbriga) - - * Correct manager variable 'EventList' case. - (Closes issue #17520. Reported, patched by kobaz) - - In addition, changes to res_timing_pthread that should make it more stable have - also been implemented. - - For a full list of changes in the current release, please see the - ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10 * Wed Jul 14 2010 Jeffrey C. Ollie - 1.6.2.8-0.3.rc1 - Add patch to remove requirement on latex2html * Tue Jun 01 2010 Marcela Maslanova - 1.6.2.8-0.2.rc1 - Mass rebuild with perl-5.12.0 * Tue May 4 2010 Jeffrey C. Ollie - 1.6.2.7-1 - * Fix building CDR and CEL SQLite3 modules. - (Closes issue #17017. Reported by alephlg. Patched by seanbright) - - * Resolve crash in SLAtrunk when the specified trunk doesn't exist. - (Reported in #asterisk-dev by philipp64. Patched by seanbright) - - * Include an extra newline after "Aliased CLI command" to get back the prompt. - (Issue #16978. Reported by jw-asterisk. Tested, patched by seanbright) - - * Prevent segfault if bad magic number is encountered. - (Issue #17037. Reported, patched by alecdavis) - - * Update code to reflect that handle_speechset has 4 arguments. - (Closes issue #17093. Reported, patched by gpatri. Tested by pabelanger, - mmichelson) - - * Resolve a deadlock in chan_local. - (Closes issue #16840. Reported, patched by bzing2, russell. Tested by bzing2) * Mon May 3 2010 Jeffrey C. Ollie - 1.6.2.7-0.2.rc3 - Update to 1.6.2.7-rc3 * Thu Apr 15 2010 Jeffrey C. Ollie - 1.6.2.7-0.1.rc2 - Update to 1.6.2.7-rc2 * Fri Mar 12 2010 Jeffrey C. Ollie - 1.6.2.6-1 - Update to final 1.6.2.6 - - The following are a few of the issues resolved by community developers: - - * Make sure to clear red alarm after polarity reversal. - (Closes issue #14163. Reported, patched by jedi98. Tested by mattbrown, - Chainsaw, mikeeccleston) - - * Fix problem with duplicate TXREQ packets in chan_iax2 - (Closes issue #16904. Reported, patched by rain. Tested by rain, dvossel) - - * Fix crash in app_voicemail related to message counting. - (Closes issue #16921. Reported, tested by whardier. Patched by seanbright) - - * Overlap receiving: Automatically send CALL PROCEEDING when dialplan starts - (Reported, Patched, and Tested by alecdavis) - - * For T.38 reINVITEs treat a 606 the same as a 488. - (Closes issue #16792. Reported, patched by vrban) - - * Fix ConfBridge crash when no timing module is loaded. - (Closes issue #16471. Reported, tested by kjotte. Patched, tested by junky) - - For a full list of changes in this releases, please see the ChangeLog: - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.6 * Mon Mar 8 2010 Jeffrey C. Ollie - 1.6.2.6-0.1.rc2 - Update to 1.6.2.6-rc2 * Mon Mar 8 2010 Jeffrey C. Ollie - 1.6.2.5-2 - Add a patch that fixes CLI history when linking against external libedit. * Thu Feb 25 2010 Jeffrey C. Ollie - 1.6.2.5-1 - Update to 1.6.2.5 - - * AST-2010-002: Invalid parsing of ACL rules can compromise security * Thu Feb 18 2010 Jeffrey C. Ollie - 1.6.2.4-1 - Update to 1.6.2.4 - - * AST-2010-002: This security release is intended to raise awareness - of how it is possible to insert malicious strings into dialplans, - and to advise developers to read the best practices documents so - that they may easily avoid these dangers. * Wed Feb 3 2010 Jeffrey C. Ollie - 1.6.2.2-1 - Update to 1.6.2.2 - - * AST-2010-001: An attacker attempting to negotiate T.38 over SIP can - remotely crash Asterisk by modifying the FaxMaxDatagram field of - the SDP to contain either a negative or exceptionally large value. - The same crash occurs when the FaxMaxDatagram field is omitted from - the SDP as well. * Fri Jan 15 2010 Jeffrey C. Ollie - 1.6.2.1-1 - Update to 1.6.2.1 final: - - * CLI 'queue show' formatting fix. - (Closes issue #16078. Reported by RoadKill. Tested by dvossel. Patched by - ppyy.) - - * Fix misreverting from 177158. - (Closes issue #15725. Reported, Tested by shanermn. Patched by dimas.) - - * Fixes subscriptions being lost after 'module reload'. - (Closes issue #16093. Reported by jlaroff. Patched by dvossel.) - - * app_queue segfaults if realtime field uniqueid is NULL - (Closes issue #16385. Reported, Tested, Patched by haakon.) - - * Fix to Monitor which previously assumed the file to write to did not contain - pathing. - (Closes issue #16377, #16376. Reported by bcnit. Patched by dant. * Tue Jan 12 2010 Jeffrey C. Ollie - 1.6.2.1-0.1.rc1 - Update to 1.6.2.1-rc1 * Sat Dec 19 2009 Jeffrey C. Ollie - 1.6.2.0-1 - Released version of 1.6.2.0 * Wed Dec 9 2009 Jeffrey C. Ollie - 1.6.2.0-0.16.rc8 - Update to 1.6.2.0-rc8 * Wed Dec 2 2009 Jeffrey C. Ollie - 1.6.2.0-0.15.rc7 - Update to 1.6.2.0-rc7 * Tue Dec 1 2009 Jeffrey C. Ollie - 1.6.2.0-0.14.rc6 - Change the logrotate and the init scripts so that Asterisk doesn't try and write to / or /root * Thu Nov 19 2009 Jeffrey C. Ollie - 1.6.2.0-0.13.rc6 - Make dependency on uw-imap conditional and some other changes to make building on RHEL5 easier. * Fri Nov 13 2009 Jeffrey C. Ollie - 1.6.2.0-0.12.rc6 - Update to 1.6.2.0-rc6 * Mon Nov 9 2009 Jeffrey C. Ollie - 1.6.2.0-0.11.rc5 - Update to 1.6.2.0-rc5 * Fri Nov 6 2009 Jeffrey C. Ollie - 1.6.2.0-0.10.rc4 - Update to 1.6.2.0-rc4 * Tue Oct 27 2009 Jeffrey C. Ollie - 1.6.2.0-0.9.rc3 - Add patch from upstream to fix how res_http_post forms paths. * Sat Oct 24 2009 Jeffrey C. Ollie - 1.6.2.0-0.8.rc3 - Add an AST_EXTRA_ARGS option to the init script - have the init script to cd to /var/spool/asterisk to prevent annoying message * Sat Oct 24 2009 Jeffrey C. Ollie - 1.6.2.0-0.7.rc3 - Compile against gmime 2.2 instead of gmime 2.4 because the patch to convert the API calls from 2.2 to 2.4 caused crashes. * Fri Oct 9 2009 Jeffrey C. Ollie - 1.6.2.0-0.6.rc3 - Require latex2html used in static-http documents * Wed Oct 7 2009 Jeffrey C. Ollie - 1.6.2.0-0.5.rc3 - Change ownership and permissions on config files to protect them. * Tue Oct 6 2009 Jeffrey C. Ollie - 1.6.2.0-0.4.rc3 - Update to 1.6.2.0-rc3 * Wed Sep 30 2009 Jeffrey C. Ollie - 1.6.2.0-0.3.rc2 - Merge firmware subpackage back into the main package. * Wed Sep 30 2009 Jeffrey C. Ollie - 1.6.2.0-0.2.rc2 - Package internal help. - Fix up some more paths in the configs so that everything ends up where we want them. * Wed Sep 30 2009 Jeffrey C. Ollie - 1.6.2.0-0.1.rc2 - Update to 1.6.2.0-rc2 - We no longer need to strip the tarball as it no longer includes non-free items. * Wed Sep 9 2009 Jeffrey C. Ollie - 1.6.1.6-2 - Enable building of API docs. - Depend on version 1.2 or newer of speex * Sun Sep 6 2009 Jeffrey C. Ollie - 1.6.1.6-1 - Update to 1.6.1.6 - Drop patches that are too troublesome to maintain anymore or have been integrated upstream. * Tue Sep 1 2009 Jeffrey C. Ollie - 1.6.1-0.26.rc1 - Add a patch from Quentin Armitage and rebuld. * Fri Aug 21 2009 Tomas Mraz - 1.6.1-0.25.rc1 - rebuilt with new openssl * Fri Jul 24 2009 Fedora Release Engineering - 1.6.1-0.24.rc1 - Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild * Thu Mar 5 2009 Jeffrey C. Ollie - 1.6.1-0.23.rc1 - Rebuild to pick up new AIS and ODBC deps. - Update script that strips out bad content from tarball to do the download and to check the GPG signature. * Mon Feb 23 2009 Fedora Release Engineering - 1.6.1-0.22.rc1 - Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild * Sun Feb 8 2009 Jeffrey C. Ollie - 1.6.1-0.21.rc1 - Update to 1.6.1-rc1 - Add backport of conference bridging that is slated for 1.6.2 - Add patches to conference bridging that implement CLI apps * Thu Jan 15 2009 Tomas Mraz - 1.6.1-0.13.beta4 - rebuild with new openssl * Sun Jan 4 2009 Jeffrey C. Ollie - 1.6.1-0.12.beta4 - Fedora Directory Server compatibility patch/subpackage. * Sun Jan 4 2009 Jeffrey C. Ollie - 1.6.1-0.10.beta4 - Fix up paths. BZ#477238 * Sat Jan 3 2009 Jeffrey C. Ollie - 1.6.1-0.9.beta4 - Update patches * Sat Jan 3 2009 Jeffrey C. Ollie - 1.6.1-0.8.beta4 - Update to 1.6.1-beta4 * Tue Dec 9 2008 Jeffrey C. Ollie - 1.6.1-0.7.beta3 - Update to 1.6.1-beta3 * Tue Dec 9 2008 Alex Lancaster - 1.6.1-0.6.beta2 - Rebuild for new gmime * Fri Nov 7 2008 Jeffrey C. Ollie - 1.6.1-0.5.beta2 - Add patch to fix missing variable on PPC. * Fri Nov 7 2008 Jeffrey C. Ollie - 1.6.1-0.4.beta2 - Update PPC systems don't have sys/io.h patch. * Fri Nov 7 2008 Jeffrey C. Ollie - 1.6.1-0.3.beta2 - PPC systems don't have sys/io.h * Fri Nov 7 2008 Jeffrey C. Ollie - 1.6.1-0.2.beta2 - Update to 1.6.1 beta 2 * Wed Nov 5 2008 Jeffrey C. Ollie - 1.6.0.1-3 - Fix issue with init script giving wrong path to config file. * Thu Oct 16 2008 Jeffrey C. Ollie - 1.6.0.1-2 - Explicitly require dahdi-tools-libs to see if we can get this to build. * Fri Oct 10 2008 Jeffrey C. Ollie - 1.6.0-1 - Update to final release. * Thu Sep 11 2008 - Bastien Nocera - 1.6.0-0.22.beta9 - Rebuild * Wed Jul 30 2008 Jeffrey C. Ollie - 1.6.0-0.21.beta9 - Replace app_rxfax/app_txfax with app_fax taken from upstream SVN. * Tue Jul 29 2008 Jeffrey C. Ollie - 1.6.0-0.20.beta9 - Bump release and rebuild with new libpri and zaptel. * Fri Jul 25 2008 Jeffrey C. Ollie - 1.6.0-0.19.beta9 - Add patch pulled from upstream SVN that fixes AST-2008-010 and AST-2008-011. * Fri Jul 25 2008 Jeffrey C. Ollie - 1.6.0-0.18.beta9 - Add patch for LDAP extracted from upstream SVN (#442011) * Thu Jul 2 2008 Jeffrey C. Ollie - 1.6.0-0.17.beta9 - Add patch that unbreaks cdr_tds with FreeTDS 0.82. - Properly obsolete conference subpackage. * Thu Jun 12 2008 Jeffrey C. Ollie - 1.6.0-0.16.beta9 - Disable building cdr_tds since new FreeTDS in rawhide no longer provides needed library. * Wed Jun 11 2008 Jeffrey C. Ollie - 1.6.0-0.15.beta9 - Bump release and rebuild to fix libtds breakage. * Mon May 19 2008 Jeffrey C. Ollie - 1.6.0-0.14.beta9 - Update to 1.6.0-beta9. - Update patches so that they apply cleanly. - Temporarily disable app_conference patch as it doesn't compile - config/scripts/postgres_cdr.sql has been merged into realtime_pgsql.sql - Re-add the asterisk-strip.sh script as a source file. * Tue Apr 22 2008 Jeffrey C. Ollie - 1.6.0-0.13.beta8 - Update to 1.6.0-beta8 - Contains fixes for AST-2008-006 / CVE-2008-1897 * Wed Apr 2 2008 Jeffrey C. Ollie - 1.6.0-0.12.beta7.1 - Return to stripped tarballs since there's more non-free content in the Asterisk tarballs than I thought. * Sun Mar 30 2008 Jeffrey C. Ollie - 1.6.0-0.11.beta7.1 - Update to 1.6.0-beta7.1 - Update patches - Back out some changes that were made because beta7 was tagged from the wrong branch. * Fri Mar 28 2008 Jeffrey C. Ollie - 1.6.0-0.10.beta7 - Update to 1.6.0-beta7 - The Asterisk tarball no longer contains the iLBC code, so we can directly use the upstream tarball without having to modify it. - Get rid of the asterisk-strip.sh script since it's no longer needed. - Diable build of codec_ilbc and format_ilbc (these do not contain any legally suspect code so they can be included in the tarball but it's pointless building them). - Update chan_mobile patch to fix API breakages. - Add a patch to chan_usbradio to fix API breakages. * Thu Mar 27 2008 Jeffrey C. Ollie - 1.6.0-0.9.beta6 - Add Postgresql schemas from contrib as documentation to the Postgresql subpackage. * Tue Mar 25 2008 Jeffrey C. Ollie - 1.6.0-0.8.beta6 - Update patches. - Add patch to compile against external libedit rather than using the in-tree version. - Add -Werror-implicit-function-declaration to optflags. - Get rid of hashtest and hashtest2 binaries that link to unfortified versions of *printf functions. They are compiled with -O0 which somehow pulls in the wrong versions. These programs aren't necessary to the operation of the package anyway. * Wed Mar 19 2008 Jeffrey C. Ollie - 1.6.0-0.6.beta6 - Update to 1.6.0-beta6 to fix some security issues. - - AST-2008-002 details two buffer overflows that were discovered in - RTP codec payload type handling. - * http://downloads.digium.com/pub/security/AST-2008-002.pdf - * All users of SIP in Asterisk 1.4 and 1.6 are affected. - - AST-2008-003 details a vulnerability which allows an attacker to - bypass SIP authentication and to make a call into the context - specified in the general section of sip.conf. - * http://downloads.digium.com/pub/security/AST-2008-003.pdf - * All users of SIP in Asterisk 1.0, 1.2, 1.4, or 1.6 are affected. - - AST-2008-004 Logging messages displayed using the ast_verbose - logging API call are not displayed as a character string, they are - displayed as a format string. - * http://downloads.digium.com/pub/security/AST-2008-004.pdf - - AST-2008-005 details a problem in the way manager IDs are caculated. - * http://downloads.digium.com/pub/security/AST-2008-005.pdf * Tue Mar 18 2008 Tom "spot" Callaway - 1.6.0-0.5.beta5 - add Requires for versioned perl (libperl.so) * Wed Mar 5 2008 Jeffrey C. Ollie - 1.6.0-0.4.beta5 - Update to 1.6.0-beta5 - Remove upstreamed patches. * Mon Mar 3 2008 Jeffrey C. Ollie - 1.6.0-0.3.beta4 - Package the directory used to store monitor recordings. * Tue Feb 26 2008 Jeffrey C. Ollie - 1.6.0-0.2.beta4 - Add patch from David Woodhouse that fixes building on PPC64. * Tue Feb 26 2008 Jeffrey C. Ollie - 1.6.0-0.1.beta4 - Update to 1.6.0 beta 4 * Wed Feb 13 2008 Jeffrey C. Ollie - 1.4.18-1 - Update to 1.4.18. - Use -march=i486 on i386 builds for atomic operations (GCC 4.3 compatibility). - Use "logger reload" instead of "logger rotate" in logrotate file (#432197). - Don't explicitly specify a group in in the init script to prevent Zaptel breakage (#426629). - Split app_ices out to a separate package so that the ices package can be required. - pbx_kdeconsole has been dropped, don't specifically exclude it from the build anymore. - Update app_conference patch. - Drop upstreamed libcap patch. * Wed Jan 2 2008 Jeffrey C. Ollie - 1.4.17-1 - Update to 1.4.17 to fix AST-2008-001. * Fri Dec 28 2007 Jeffrey C. Ollie - 1.4.16.2-1 - Update to 1.4.16.2 * Sat Dec 22 2007 Jeffrey C. Ollie - 1.4.16.1-2 - Bump release and rebuild to fix broken dep on uw-imap. * Wed Dec 19 2007 Jeffrey C. Ollie - 1.4.16.1-1 - Update to the real 1.4.16.1. * Wed Dec 19 2007 Jeffrey C. Ollie - 1.4.16-2 - Add patch to bring source up to version 1.4.16.1 which will be released shortly to fix some crasher bugs introduced by 1.4.16. * Tue Dec 18 2007 Jeffrey C. Ollie - 1.4.16-1 - Update to 1.4.16 to fix security bug. * Sat Dec 15 2007 Jeffrey C. Ollie - 1.4.15-7 - Really, really fix the build problems on devel. * Sat Dec 15 2007 Jeffrey C. Ollie - 1.4.15-6 - Tweaks to get to build on x86_64 * Wed Dec 12 2007 Jeffrey C. Ollie - 1.4.15-5 - Exclude PPC64 * Wed Dec 12 2007 Jeffrey C. Ollie - 1.4.15-4 - Don't build apidocs by default since there's a problem building on x86_64. * Tue Dec 11 2007 Jeffrey C. Ollie - 1.4.15-3 - Really get rid of zero length map files. * Mon Dec 10 2007 Jeffrey C. Ollie - 1.4.15-2 - Get rid of zero length map files. - Shorten descriptions of voicemail subpackages * Fri Nov 30 2007 Jeffrey C. Ollie - 1.4.15-1 - Update to 1.4.15 * Tue Nov 20 2007 Jeffrey C. Ollie - 1.4.14-2 - Fix license and other rpmlint warnings. * Mon Nov 19 2007 Jeffrey C. Ollie - 1.4.14-1 - Update to 1.4.14 * Fri Nov 16 2007 Jeffrey C. Ollie - 1.4.13-7 - Add chan_mobile * Tue Nov 13 2007 Jeffrey C. Ollie - 1.4.13-6 - Don't build cdr_sqlite because sqlite2 has been orphaned. - Rebase local patches to latest upstream SVN - Update app_conference patch to latest from upstream SVN - Apply post-1.4.13 patches from upstream SVN * Wed Oct 10 2007 Jeffrey C. Ollie - 1.4.13-1 - Update to 1.4.13 * Tue Oct 9 2007 Jeffrey C. Ollie - 1.4.12.1-1 - Update to 1.4.12.1 * Wed Aug 22 2007 Jeffrey C. Ollie - 1.4.11-1 - Update to 1.4.11 * Fri Aug 10 2007 Jeffrey C. Ollie - 1.4.10.1-1 - Update to 1.4.10.1. * Tue Aug 7 2007 Jeffrey C. Ollie - 1.4.10-1 - Update to 1.4.10 (security update). * Tue Aug 7 2007 Jeffrey C. Ollie - 1.4.9-7 - Add a patch that allows alternate extensions to be defined in users.conf * Mon Aug 6 2007 Jeffrey C. Ollie - 1.4.9-6 - Update app_conference patch. Enter/leave sounds are now possible. * Fri Jul 27 2007 Jeffrey C. Ollie - 1.4.9-5 - Update patches so we don't need to run auto* tools, because autoconf 2.60 is required and FC-6 and RHEL5 only have autoconf 2.59. * Thu Jul 26 2007 Jeffrey C. Ollie - 1.4.9-4 - Don't build app_mp3 * Wed Jul 25 2007 Jeffrey C. Ollie - 1.4.9-3 - Add app_conference * Wed Jul 25 2007 Jeffrey C. Ollie - 1.4.9-2 - Use plain useradd/groupadd rather than the fedora-usermgmt - Clean up requirements - Clean up build requirements by moving them to package sections * Tue Jul 24 2007 Jeffrey C. Ollie - 1.4.9-1 - Update to 1.4.9 * Tue Jul 17 2007 Jeffrey C. Ollie - 1.4.8-1 - Update to 1.4.8 - Drop ixjuser patch. * Tue Jul 10 2007 Jeffrey C. Ollie - 1.4.7.1-1 - Update to 1.4.7.1 * Mon Jul 9 2007 Jeffrey C. Ollie - 1.4.7-1 - Update to 1.4.7 - RxFAX/TxFAX applications * Sun Jul 1 2007 Jeffrey C. Ollie - 1.4.6-4 - It's "sbin", not "bin" silly. * Sat Jun 30 2007 Jeffrey C. Ollie - 1.4.6-3 - Add patch that lets us change TOS bits even when running non-root * Fri Jun 29 2007 Jeffrey C. Ollie - 1.4.6-2 - voicemail needs to require /usr/bin/sox and /usr/bin/sendmail * Fri Jun 29 2007 Jeffrey C. Ollie - 1.4.6-1 - Update to 1.4.6 - Remove upstreamed patch. * Thu Jun 21 2007 Jeffrey C. Ollie - 1.4.5-10 - Build the IMAP and ODBC storage options of voicemail and split voicemail out into subpackages. - Apply patch so that the system UW IMAP libray can be linked against. - Patch modules.conf.sample so that alternal voicemail modules don't get loaded simultaneously. - Link against system GSM library rather than internal copy. - Patch the Makefile so that it doesn't add redundant/wrong compiler options. - Force building with the standard RPM optimization flags. - Install the Asterisk MIB in a location that net-snmp can find it. - Only package docs in the main package that are relevant and that haven't been packaged by a subpackage. - Other minor cleanups. * Mon Jun 18 2007 Jeffrey C. Ollie - 1.4.5-9 - Move sounds * Mon Jun 18 2007 Jeffrey C. Ollie - 1.4.5-8 - Update some more ownership/permissions * Mon Jun 18 2007 Jeffrey C. Ollie - 1.4.5-7 - Fix some permissions. * Mon Jun 18 2007 Jeffrey C. Ollie - 1.4.5-6 - Update init script patch - Move pid file to subdir of /var/run * Mon Jun 18 2007 Jeffrey C. Ollie - 1.4.5-5 - Update init script patch to run as non-root * Sun Jun 17 2007 Jeffrey C. Ollie - 1.4.5-4 - Build modules that depend on FreeTDS. - Don't build voicemail with ODBC storage. * Sun Jun 17 2007 Jeffrey C. Ollie - 1.4.5-3 - Have the build output the commands executing, rather than covering them up. * Fri Jun 15 2007 Jeffrey C. Ollie - 1.4.5-1 - Update to 1.4.5 - Remove upstreamed patch. * Wed May 9 2007 Jeffrey C. Ollie - 1.4.4-2 - Add a patch to fix CVE-2007-2488/ASA-2007-013 * Fri Apr 27 2007 Jeffrey C. Ollie - 1.4.4-1 - Update to 1.4.4 * Wed Mar 21 2007 Jeffrey C. Ollie - 1.4.2-1 - Update to 1.4.2 * Tue Mar 6 2007 Jeffrey C. Ollie - 1.4.1-2 - Package the IAXy firmware - Minor clean-ups in files * Mon Mar 5 2007 Jeffrey C. Ollie - 1.4.1-1 - Update to 1.4.1 - Don't build/package codec_zap (zaptel 1.4.0 doesn't support it) * Fri Dec 15 2006 Jeffrey C. Ollie - 1.4.0-6.beta4 - Update to 1.4.0-beta4 - Various cleanups. * Fri Oct 20 2006 Jeffrey C. Ollie - 1.4.0-5.beta3 - Don't package IAXy firmware because of license - Don't build app_rpt - Don't BR lm_sensors on PPC - Better way to prevent download/installation of sound archives - Redo tarball to eliminate non-free items * Thu Oct 19 2006 Jeffrey C. Ollie - 1.4.0-4.beta3 - Remove explicit dependency on glibc-kernheaders. - Build jabber modules on PPC * Wed Oct 18 2006 Jeffrey C. Ollie - 1.4.0-3.beta3 - *Really* update to beta3 - chan_jingle has been taken out of 1.4 - Move misplaced binaries to where they should be * Wed Oct 18 2006 Jeffrey C. Ollie - 1.4.0-2.beta3 - Remove requirement on asterisk-sounds-core until licensing can be figured out. * Wed Oct 18 2006 Jeffrey C. Ollie - 1.4.0-1.beta3 - Update to 1.4.0-beta3 * Sun Oct 15 2006 Jeffrey C. Ollie - 1.4.0-0.beta2 - Update to 1.4.0-beta2 * Tue Jul 25 2006 Jeffrey C. Ollie - 1.2.10-1 - Update to 1.2.10. * Wed Jun 7 2006 Jeffrey C. Ollie - 1.2.9.1 - Update to 1.2.9.1 * Fri Jun 2 2006 Jeffrey C. Ollie - 1.2.8 - Update to 1.2.8 - Add misdn.conf to list of configs. - Drop chan_bluetooth patch for now... * Tue May 2 2006 Jeffrey C. Ollie - 1.2.7.1-6 - Zaptel subpackage shouldn't obsolete the sqlite subpackage. - Remove mISDN until build issues can be figured out. * Mon Apr 24 2006 Jeffrey C. Ollie - 1.2.7.1-5 - Build mISDN channel drivers, modelled after spec file from David Woodhouse * Thu Apr 20 2006 Jeffrey C. Ollie - 1.2.7.1-4 - Update chan_bluetooth patch with some additional information as to it's source and comment out more in the configuration file. * Thu Apr 20 2006 Jeffrey C. Ollie - 1.2.7.1-3 - Add chan_bluetooth * Wed Apr 19 2006 Jeffrey C. Ollie - 1.2.7.1-2 - Split off more stuff into subpackages. * Wed Apr 12 2006 Jeffrey C. Ollie - 1.2.7-1 - Update to 1.2.7 * Mon Apr 10 2006 Jeffrey C. Ollie - 1.2.6-3 - Fix detection of libpri on 64 bit arches (taken from Matthias Saou's rpmforge package) - Change sqlite subpackage name to sqlite2 (there are sqlite3 modules in development). * Thu Apr 6 2006 Jeffrey C. Ollie - 1.2.6-2 - Don't build GTK 1.X console since GTK 1.X is being moved out of core... * Mon Mar 27 2006 Jeffrey C. Ollie - 1.2.6-1 - Update to 1.2.6 * Mon Mar 6 2006 Jeffrey C. Ollie - 1.2.5-1 - Update to 1.2.5. - Removed upstreamed MOH patch. - Add full urls to the app_(r|t)xfax.c sources. - Update spandsp patch. * Mon Feb 13 2006 Jeffrey C. Ollie - 1.2.4-4 - Actually apply the patch. * Mon Feb 13 2006 Jeffrey C. Ollie - 1.2.4-3 - Add patch to keep Asterisk from crashing when using MOH inside a MeetMe conference. * Mon Feb 6 2006 Jeffrey C. Ollie - 1.2.4-2 - BR sqlite2-devel * Tue Jan 31 2006 Jeffrey C. Ollie - 1.2.4-1 - Update to 1.2.4. * Wed Jan 25 2006 Jeffrey C. Ollie - 1.2.3-4 - Took some tricks from Asterisk packages by Roy-Magne Mo. - Enable gtk console module. - BR gtk+-devel. - Add logrotate script. - BR sqlite2-devel and new sqlite subpackage. - BR doxygen and graphviz for building duxygen documentation. (But don't build it yet.) * Wed Jan 25 2006 Jeffrey C. Ollie - 1.2.3-3 - Completely eliminate the "asterisk" user from the spec file. - Move more config files to subpackages. - Consolidate two patches that patch the init script. - BR curl-devel - BR alsa-lib-devel - alsa, curl, oss subpackages * Wed Jan 25 2006 Jeffrey C. Ollie - 1.2.3-2 - Do not run as user "asterisk" as that prevents setting of IP TOS (which is bad for quality of service). - Add patch for setting TOS separately for SIP and RTP packets. * Wed Jan 25 2006 Jeffrey C. Ollie - 1.2.3-1 - First version for Fedora Extras.