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File asterisk-17.9.2.spec of Package asterisk
#global _rc 2 #global _beta 3 %define debug_package %{nil} %{!?__global_ldflags: %global __global_ldflags -Wl,-z,relro} %global _smp_mflags -j1 %global optflags %{optflags} -Werror-implicit-function-declaration -DLUA_COMPAT_MODULE %ifarch s390 %{arm} aarch64 %global ldflags -Wl,--as-needed,--library-path=%{_libdir} %{__global_ldflags} %else %global ldflags -m%{__isa_bits} -Wl,--as-needed,--library-path=%{_libdir} %{__global_ldflags} -fPIC %endif %if 0%{?fedora} >= 15 || 0%{?rhel} >= 7 %global astvarrundir /run/asterisk %global tmpfilesd 1 %else %global astvarrundir %{_localstatedir}/run/asterisk %global tmpfilesd 0 %endif %if 0%{?fedora} >= 16 || 0%{?rhel} >= 7 %global systemd 1 %else %global systemd 0 %endif %global apidoc 0 %global mysql 1 %global odbc 1 %global postgresql 1 %global radius 1 %global snmp 1 %if 0%{?fedora} >= 21 || 0%{?rhel} >= 8 %global misdn 0 %else %global misdn 1 %endif %global ldap 1 %if 0%{?rhel} >= 8 %global gmime 0 %global radius 0 %global corosync 0 %else %global gmime 1 %global radius 1 %global corosync 1 %endif %if 0%{?fedora} <= 24 %global gmime 0 %global radius 0 %global mysql 0 %endif %if 0%{?fedora} >= 21 || 0%{?rhel} >= 8 %global jack 0 %else %global jack 1 %endif %global xmpp 0 %global makeargs DEBUG= OPTIMIZE= DESTDIR=%{buildroot} ASTVARRUNDIR=%{astvarrundir} ASTDATADIR=%{_datadir}/asterisk ASTVARLIBDIR=%{_datadir}/asterisk ASTDBDIR=%{_localstatedir}/spool/asterisk NOISY_BUILD=1 Summary: The Open Source PBX Name: asterisk Version: 17.9.2 Release: <RELEASE>%{?_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: asterisk17-logrotate Source3: asterisk17-menuselect.makedeps Source4: asterisk17-menuselect.makeopts Source5: asterisk17.service Source6: asterisk17-tmpfiles Source7: asterisk-addons-mp3-rev202.tar.gz Source8: pjproject-2.10.tar.bz2 Source9: pjproject-2.10.md5 Source10: jansson-2.12.tar.bz2 Source98: modules.conf Source99: indications.conf %if 0%{?fedora} > 0 Patch0: gcc5.rc2.patch %endif Patch1: asterisk17-init-path.patch Patch2: asterisk-addons-mp3.patch %if 0%{?rhel} >= 8 Patch3: asterisk17-python-to-python2.patch %endif 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 %if 0%{?gmime} BuildRequires: gtk2-devel %endif BuildRequires: libsrtp-devel BuildRequires: popt-devel %if %{systemd} BuildRequires: systemd-units %endif # for res_http_post %if (0%{?fedora} > 0 || 0%{?rhel} >= 7) && 0%{?gmime} BuildRequires: gmime-devel %endif # for building docs BuildRequires: doxygen BuildRequires: graphviz BuildRequires: libxml2-devel BuildRequires: latex2html # for building res_calendar_caldav BuildRequires: neon-devel BuildRequires: libical-devel BuildRequires: libxml2-devel # for codec_speex BuildRequires: speex-devel >= 1.2 %if (0%{?fedora} > 0 || 0%{?rhel} >= 8) BuildRequires: speexdsp-devel >= 1.2 %endif # for format_ogg_vorbis BuildRequires: libogg-devel BuildRequires: libvorbis-devel # codec_gsm BuildRequires: gsm-devel # additional dependencies BuildRequires: SDL-devel #BuildRequires: SDL_image-devel # cli BuildRequires: libedit-devel # codec_ilbc BuildRequires: ilbc-devel # res_rtp_asterisk BuildRequires: libuuid-devel %if 0%{?corosync} BuildRequires: corosynclib-devel %endif BuildRequires: alsa-lib-devel BuildRequires: libcurl-devel 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 BuildRequires: spandsp-devel >= 0.0.5-0.1.pre4 BuildRequires: libtiff-devel BuildRequires: libjpeg-devel BuildRequires: lua-devel %if 0%{?jack} BuildRequires: jack-audio-connection-kit-devel %endif BuildRequires: libresample-devel BuildRequires: bluez-libs-devel BuildRequires: libtool-ltdl-devel BuildRequires: portaudio-devel >= 19 BuildRequires: sqlite-devel BuildRequires: freetds-devel %if 0%{?misdn} BuildRequires: mISDN-devel %endif %if 0%{?ldap} BuildRequires: openldap-devel %endif %if 0%{?mysql} %if 0%{?rhel} >= 7 BuildRequires: mariadb-devel %else BuildRequires: mariadb-connector-c-devel %endif %endif %if 0%{?odbc} BuildRequires: libtool-ltdl-devel BuildRequires: unixODBC-devel %endif %if 0%{?postgresql} BuildRequires: postgresql-devel %endif %if 0%{?radius} BuildRequires: radiusclient-ng-devel %endif %if 0%{?snmp} BuildRequires: net-snmp-devel BuildRequires: lm_sensors-devel %endif %if 0%{?fedora} > 0 || 0%{?rhel} >= 7 BuildRequires: uw-imap-devel BuildRequires: openssl-devel %endif #BuildRequires: pjproject-devel #BuildRequires: jansson-devel BuildRequires: fftw-devel #BuildRequires: xmlstarlet Requires(pre): %{_sbindir}/useradd Requires(pre): %{_sbindir}/groupadd %if 0%{?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 been removed in 10.4.0 Obsoletes: asterisk-usbradio <= 10.3.1-1 # If upgrading from digium releases we need to handle the package format changes Obsoletes: asterisk-core <= %{version}-%{release} Conflicts: asterisk-core <= %{version}-%{release} Provides: asterisk-core = %{version}-%{release} Obsoletes: asterisk-configs <= %{version}-%{release} Conflicts: asterisk-configs <= %{version}-%{release} Provides: asterisk-configs = %{version}-%{release} Obsoletes: asterisk-doc <= %{version}-%{release} Conflicts: asterisk-doc <= %{version}-%{release} Provides: asterisk-doc = %{version}-%{release} %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. %package ael Summary: AEL (Asterisk Extension Logic) modules for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description ael AEL (Asterisk Extension Logic) mdoules for Asterisk %package alsa Summary: Modules for Asterisk that use Alsa sound drivers Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description alsa Modules for Asterisk that use Alsa sound drivers. %if 0%{?apidoc} %package apidoc Summary: API documentation for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description apidoc API documentation for Asterisk. %endif %package calendar Summary: Calendar applications for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description calendar Calendar applications for Asterisk. %package compat Summary: Metapackage to help transition Asterisk users to the new package split Obsoletes: asterisk < 13.0.0 Requires: asterisk = %{version}-%{release} Requires: asterisk-ael = %{version}-%{release} Requires: asterisk-iax2 = %{version}-%{release} Requires: asterisk-mgcp = %{version}-%{release} Requires: asterisk-phone = %{version}-%{release} Requires: asterisk-sip = %{version}-%{release} %description compat This package only exists to help transition Asterisk users to the new package split. It will be removed after one distribution release cycle, please do not reference it or depend on it in any way. %if 0%{?corosync} %package corosync Summary: Modules for Asterisk that use Corosync Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description corosync Modules for Asterisk that use Corosync. %endif %package curl Summary: Modules for Asterisk that use cURL Group: Applications/Internet Requires: asterisk = %{version}-%{release} %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 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} %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. %package iax2 Summary: IAX2 channel driver for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description iax2 IAX2 channel driver for Asterisk %package hep Summary: Modules for capturing SIP traffic using Homer (HEPv3) Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description hep Modules for capturing SIP traffic using Homer (HEPv3) %if 0%{?fedora} || 0%{?rhel} >= 7 %package ices Summary: Stream audio from Asterisk to an IceCast server Group: Applications/Internet Requires: asterisk = %{version}-%{release} Requires: ices %description ices Stream audio from Asterisk to an IceCast server. %endif %if 0%{?jack} %package jack Summary: JACK resources for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description jack JACK resources for Asterisk. %endif %package lua Summary: Lua resources for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description lua Lua resources for Asterisk. %if 0%{?ldap} %package ldap Summary: LDAP resources for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description ldap LDAP resources for Asterisk. %endif %if 0%{?misdn} %package misdn Summary: mISDN channel for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} Requires(pre): %{_sbindir}/usermod %description misdn mISDN channel for Asterisk. %endif %package mgcp Summary: MGCP channel driver for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description mgcp MGCP channel driver for Asterisk %package mobile Summary: Mobile (BlueTooth) channel for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} Requires(pre): %{_sbindir}/usermod Obsoletes: asterisk-addons-bluetooth <= %{version}-%{release} Provides: asterisk-addons-bluetooth = %{version}-%{release} Conflicts: asterisk-addons-bluetooth <= %{version}-%{release} %description mobile Mobile (BlueTooth) channel for Asterisk. %package mp3 Summary: MP3 format support for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description mp3 MP3 format support for Asterisk (format_mp3 addon) %package minivm Summary: MiniVM applicaton for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description minivm MiniVM application for Asterisk. %package mwi-external Summary: Support for developing external voicemail applications Group: Applications/Internet Requires: asterisk = %{version}-%{release} Conflicts: asterisk-voicemail = %{version}-%{release} Conflicts: asterisk-voicemail-implementation = %{version}-%{release} %description mwi-external Support for developing external voicemail applications %if 0%{?mysql} %package mysql Summary: Applications for Asterisk that use MySQL Group: Applications/Internet Requires: asterisk = %{version}-%{release} Obsoletes: asterisk-addons-mysql <= %{version}-%{release} Provides: asterisk-addons-mysql = %{version}-%{release} Conflicts: asterisk-addons-mysql <= %{version}-%{release} %description mysql Applications for Asterisk that use MySQL. %endif %if 0%{?odbc} %package odbc Summary: Applications for Asterisk that use ODBC (except voicemail) Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description odbc Applications for Asterisk that use ODBC (except voicemail) %endif %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} Obsoletes: asterisk-addons-ooh323 <= %{version}-%{release} Provides: asterisk-addons-ooh323 = %{version}-%{release} Conflicts: asterisk-addons-ooh323 <= %{version}-%{release} %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. %if 0%{?xmpp} %package phone Summary: Channel driver for Quicknet Technologies, Inc.'s Telephony cards Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description phone Quicknet Technologies, Inc.'s Telephony cards including the Internet PhoneJACK, Internet PhoneJACK Lite, Internet PhoneJACK PCI, Internet LineJACK, Internet PhoneCARD and SmartCABLE. %endif %package pjsip Summary: SIP channel based upon the PJSIP library Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description pjsip SIP channel based upon the PJSIP library %package portaudio Summary: Modules for Asterisk that use the portaudio library Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description portaudio Modules for Asterisk that use the portaudio library. %if 0%{?postgresql} %package postgresql Summary: Applications for Asterisk that use PostgreSQL Group: Applications/Internet Requires: asterisk = %{version}-%{release} Obsoletes: asterisk-pgsql <= %{version}-%{release} Provides: asterisk-pgsql = %{version}-%{release} Conflicts: asterisk-pgsql <= %{version}-%{release} %description postgresql Applications for Asterisk that use PostgreSQL. %endif %if 0%{?radius} %package radius Summary: Applications for Asterisk that use RADIUS Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description radius Applications for Asterisk that use RADIUS. %endif %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 sip Summary: Legacy SIP channel driver for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} %description sip Legacy SIP channel driver for Asterisk %if 0%{?snmp} %package snmp Summary: Module that enables SNMP monitoring of Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} # 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. %endif %package sqlite Summary: Sqlite modules for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} Obsoletes: asterisk-sqlite3 <= %{version}-%{release} Provides: asterisk-sqlite3 = %{version}-%{release} Conflicts: asterisk-sqlite3 <= %{version}-%{release} %description sqlite Sqlite modules for Asterisk. %package tds Summary: Modules for Asterisk that use FreeTDS Group: Applications/Internet Requires: asterisk = %{version}-%{release} %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 and local storage Group: Applications/Internet Requires: asterisk = %{version}-%{release} Provides: asterisk-voicemail-implementation = %{version}-%{release} Requires: /usr/bin/sox Requires: /usr/sbin/sendmail Conflicts: asterisk-mwi-external <= %{version}-%{release} %description voicemail Voicemail implementation for Asterisk that stores voicemail on the local filesystem. %if 0%{?fedora} > 0 || 0%{?rhel} >= 7 %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} Obsoletes: asterisk-voicemail-imapstorage <= %{version}-%{release} Provides: asterisk-voicemail-imapstorage = %{version}-%{release} Conflicts: asterisk-voicemail-imapstorage <= %{version}-%{release} %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} Obsoletes: asterisk-voicemail-odbcstorage <= %{version}-%{release} Provides: asterisk-voicemail-odbcstorage = %{version}-%{release} Conflicts: asterisk-voicemail-odbcstorage <= %{version}-%{release} %description voicemail-odbc Voicemail implementation for Asterisk that uses ODBC to store voicemail in a database. %if 0%{?xmpp} %package xmpp Summary: Jabber/XMPP resources for Asterisk Group: Applications/Internet Requires: asterisk = %{version}-%{release} Obsoletes: asterisk-jabber < 13.0.0 Conflicts: asterisk-jabber < 13.0.0 %description xmpp Jabber/XMPP resources for Asterisk. %endif %prep %setup -q -n asterisk-%{version}%{?_rc:-rc%{_rc}}%{?_beta:-beta%{_beta}} %setup -q -n asterisk-%{version}%{?_rc:-rc%{_rc}}%{?_beta:-beta%{_beta}} -T -D -a 7 #%if 0%{?fedora} > 0 #%patch0 -p1 #%endif %patch1 -p1 %patch2 -p0 %if 0%{?rhel} >= 8 %patch3 -p1 %endif 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:.*$/install:/' sounds/Makefile # convert comments in one file to UTF-8 mv main/fskmodem.c main/fskmodem.c.old iconv -f iso-8859-1 -t utf-8 -o main/fskmodem.c main/fskmodem.c.old touch -r main/fskmodem.c.old main/fskmodem.c rm main/fskmodem.c.old chmod -x contrib/scripts/dbsep.cgi %if ! 0%{?xmpp} %{__perl} -pi -e 's/^MENUSELECT_CHANNELS=(.*)$/MENUSELECT_CHANNELS=\1 chan_motif/g' menuselect.makeopts %{__perl} -pi -e 's/^MENUSELECT_CHANNELS=(.*)$/MENUSELECT_CHANNELS=\1 chan_phone/g' menuselect.makeopts %{__perl} -pi -e 's/^MENUSELECT_RES=(.*)$/MENUSELECT_RES=\1 res_xmpp/g' menuselect.makeopts %{__perl} -pi -e 's/^MENUSELECT_RES=(.*)$/MENUSELECT_RES=\1 res_calendar_exchange/g' menuselect.makeopts %endif %if 0%{?rhel} == 6 %{__perl} -pi -e 's/^MENUSELECT_RES=(.*)$/MENUSELECT_RES=\1 res_http_post/g' menuselect.makeopts %{__perl} -pi -e 's/^MENUSELECT_APPS=(.*)$/MENUSELECT_APPS=\1 app_voicemail_imap/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=(.*)$/MENUSELECT_RES=\1 res_timing_timerfd/g' menuselect.makeopts %endif %if ! 0%{?corosync} %{__perl} -pi -e 's/^MENUSELECT_RES=(.*)$/MENUSELECT_RES=\1 res_corosync/g' menuselect.makeopts %endif %if ! 0%{?mysql} %{__perl} -pi -e 's/^MENUSELECT_ADDONS=(.*)$/MENUSELECT_ADDONS=\1 res_config_mysql app_mysql cdr_mysql/g' menuselect.makeopts %endif %if ! 0%{?postgresql} %{__perl} -pi -e 's/^MENUSELECT_RES=(.*)$/MENUSELECT_RES=\1 res_config_pgsql/g' menuselect.makeopts %{__perl} -pi -e 's/^MENUSELECT_CDR=(.*)$/MENUSELECT_CDR=\1 cdr_pgsql/g' menuselect.makeopts %{__perl} -pi -e 's/^MENUSELECT_CEL=(.*)$/MENUSELECT_CEL=\1 cel_pgsql/g' menuselect.makeopts %endif %if ! 0%{?radius} %{__perl} -pi -e 's/^MENUSELECT_CDR=(.*)$/MENUSELECT_CDR=\1 cdr_radius/g' menuselect.makeopts %{__perl} -pi -e 's/^MENUSELECT_CEL=(.*)$/MENUSELECT_CEL=\1 cel_radius/g' menuselect.makeopts %endif %if ! 0%{?snmp} %{__perl} -pi -e 's/^MENUSELECT_RES=(.*)$/MENUSELECT_RES=\1 res_snmp/g' menuselect.makeopts %endif %if ! 0%{?misdn} %{__perl} -pi -e 's/^MENUSELECT_CHANNELS=(.*)$/MENUSELECT_CHANNELS=\1 chan_misdn/g' menuselect.makeopts %endif %if ! 0%{?jack} %{__perl} -pi -e 's/^MENUSELECT_APPS=(.*)$/MENUSELECT_APPS=\1 app_jack/g' menuselect.makeopts %endif %if ! 0%{?ldap} %{__perl} -pi -e 's/^MENUSELECT_RES=(.*)$/MENUSELECT_RES=\1 res_config_ldap/g' menuselect.makeopts %endif %if ! 0%{?gmime} %{__perl} -pi -e 's/^MENUSELECT_RES=(.*)$/MENUSELECT_RES=\1 res_http_post/g' menuselect.makeopts %endif %build export CFLAGS="%{optflags}" export CXXFLAGS="%{optflags}" export FFLAGS="%{optflags}" export LDFLAGS="%{ldflags}" export ASTCFLAGS=" " export JANSSON_CONFIGURE_OPTS="--with-pic" ./bootstrap.sh pushd menuselect %configure popd mkdir %{_tmppath}/dlcache cp %{SOURCE8} %{_tmppath}/dlcache/ cp %{SOURCE9} %{_tmppath}/dlcache/ cp %{SOURCE10} %{_tmppath}/dlcache/ %if 0%{?fedora} > 0 || 0%{?rhel} >= 7 %configure --with-imap=system --with-gsm=/usr --with-ilbc=/usr --with-libedit=yes --with-srtp --with-jansson-bundled --with-download-cache=%{_tmppath}/dlcache LDFLAGS="%{ldflags}" %else %configure --with-gsm=/usr --with-ilbc=/usr --with-libedit=yes --with-gmime=no --with-srtp --with-jansson-bundled --with-download-cache=%{_tmppath}/dlcache LDFLAGS="%{ldflags}" %endif make %{?_smp_mflags} menuselect-tree NOISY_BUILD=1 %{__perl} -n -i -e 'print unless /openr2/i' menuselect-tree make %{?_smp_mflags} %{makeargs} #rm apps/app_voicemail.o apps/app_directory.o #mv apps/app_voicemail.so apps/app_voicemail_plain.so #mv apps/app_directory.so apps/app_directory_plain.so #%if 0%{?fedora} > 0 || 0%{?rhel} >= 7 #sed -i -e 's/^MENUSELECT_OPTS_app_voicemail=.*$/MENUSELECT_OPTS_app_voicemail=IMAP_STORAGE/' menuselect.makeopts #make %{?_smp_mflags} %{makeargs} # #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 #make %{?_smp_mflags} %{makeargs} # #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 #touch apps/app_voicemail.o apps/app_directory.o #touch apps/app_voicemail.so apps/app_directory.so sed -i -e 's/^MENUSELECT_RES=\(.*\)\bres_mwi_external\b\(.*\)$/MENUSELECT_RES=\1 \2/g' menuselect.makeopts sed -i -e 's/^MENUSELECT_RES=\(.*\)\bres_mwi_external_ami\b\(.*\)$/MENUSELECT_RES=\1 \2/g' menuselect.makeopts sed -i -e 's/^MENUSELECT_RES=\(.*\)\bres_stasis_mailbox\b\(.*\)$/MENUSELECT_RES=\1 \2/g' menuselect.makeopts sed -i -e 's/^MENUSELECT_APP=\(.*\)$/MENUSELECT_RES=\1 app_voicemail/g' menuselect.makeopts make %{?_smp_mflags} %{makeargs} %if 0%{?apidoc} make %{?_smp_mflags} progdocs %{makeargs} # fix dates so that we don't get multilib conflicts find doc/api/html -type f -print0 | xargs --null touch -r ChangeLog %endif %install rm -rf %{buildroot} export CFLAGS="%{optflags}" export CXXFLAGS="%{optflags}" export FFLAGS="%{optflags}" export LDFLAGS="%{ldflags}" export ASTCFLAGS="%{optflags}" make install %{makeargs} make samples %{makeargs} make install-headers %{makeargs} %if 0%{?systemd} install -D -p -m 0644 %{SOURCE5} %{buildroot}%{_unitdir}/asterisk.service rm -f %{buildroot}%{_sbindir}/safe_asterisk %else mkdir -p %{buildroot}%{_initrddir} build_tools/install_subst contrib/init.d/rc.redhat.asterisk %{buildroot}%{_initrddir}/asterisk %endif install -D -p -m 0644 %{S:2} %{buildroot}%{_sysconfdir}/logrotate.d/asterisk install -D -p -m 0644 %{SOURCE98} %{buildroot}%{_sysconfdir}/asterisk/modules.conf install -D -p -m 0644 %{SOURCE99} %{buildroot}%{_sysconfdir}/asterisk/indications.conf #rm %{buildroot}%{_libdir}/asterisk/modules/app_directory.so #rm %{buildroot}%{_libdir}/asterisk/modules/app_voicemail.so %if 0%{?fedora} > 0 || 0%{?rhel} >= 7 #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.so %{buildroot}%{_libdir}/asterisk/modules/app_directory.so install -D -p -m 0755 apps/app_voicemail.so %{buildroot}%{_libdir}/asterisk/modules/app_voicemail.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 # Don't include refcounter rm -rf %{buildroot}%{_datadir}/asterisk/scripts/refcounter.* # rm -rf %{buildroot}%{_sysconfdir}/asterisk/app_skel.conf rm -rf %{buildroot}%{_sysconfdir}/asterisk/config_test.conf rm -rf %{buildroot}%{_sysconfdir}/asterisk/test_sorcery.conf %if ! 0%{?xmpp} # With iksemel gone and the xmpp package unavailable, we can't use the configs rm -rf %{buildroot}%{_sysconfdir}/asterisk/motif.conf rm -rf %{buildroot}%{_sysconfdir}/asterisk/xmpp.conf rm -rf %{buildroot}%{_sysconfdir}/asterisk/phone.conf %endif %if 0%{?apidoc} find doc/api/html -name \*.map -size 0 -delete %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 mkdir -p %{buildroot}%{astvarrundir} %endif %if ! 0%{?mysql} rm -f %{buildroot}%{_sysconfdir}/asterisk/*_mysql.conf %endif %if ! 0%{?postgresql} rm -f %{buildroot}%{_sysconfdir}/asterisk/*_pgsql.conf %endif %if ! 0%{?misdn} rm -f %{buildroot}%{_sysconfdir}/asterisk/misdn.conf %endif %if ! 0%{?snmp} rm -f %{buildroot}%{_sysconfdir}/asterisk/res_snmp.conf %endif %if ! 0%{?ldap} rm -f %{buildroot}%{_sysconfdir}/asterisk/res_ldap.conf %endif %if ! 0%{?corosync} rm -f %{buildroot}%{_sysconfdir}/asterisk/res_corosync.conf %endif rm -f %{buildroot}%{_sysconfdir}/asterisk/cdr_beanstalkd.conf rm -f %{buildroot}%{_sysconfdir}/asterisk/cel_beanstalkd.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 %if 0%{?misdn} %pre misdn %{_sbindir}/usermod -a -G misdn asterisk %endif %files %defattr(-,root,root,-) %doc README* *.txt ChangeLog BUGS CREDITS configs %doc doc/asterisk.sgml %if %{systemd} %{_unitdir}/asterisk.service %else %{_initrddir}/asterisk %endif %{_libdir}/libasteriskssl.so.1 %{_libdir}/libasteriskssl.so %{_libdir}/libasteriskpj.so.2 %{_libdir}/libasteriskpj.so %dir %{_libdir}/asterisk %dir %{_libdir}/asterisk/modules %{_libdir}/asterisk/modules/app_agent_pool.so %{_libdir}/asterisk/modules/app_adsiprog.so %{_libdir}/asterisk/modules/app_alarmreceiver.so %{_libdir}/asterisk/modules/app_amd.so %{_libdir}/asterisk/modules/app_attended_transfer.so %{_libdir}/asterisk/modules/app_authenticate.so %{_libdir}/asterisk/modules/app_blind_transfer.so %{_libdir}/asterisk/modules/app_bridgeaddchan.so %{_libdir}/asterisk/modules/app_bridgewait.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_stasis.so %{_libdir}/asterisk/modules/app_statsd.so %{_libdir}/asterisk/modules/app_stream_echo.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_builtin_interval_features.so %{_libdir}/asterisk/modules/bridge_holding.so %{_libdir}/asterisk/modules/bridge_native_rtp.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_bridge_media.so #%{_libdir}/asterisk/modules/chan_multicast_rtp.so %{_libdir}/asterisk/modules/chan_rtp.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_ilbc.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_ilbc.so #%{_libdir}/asterisk/modules/format_jpeg.so %{_libdir}/asterisk/modules/format_ogg_speex.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_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_hangupcause.so %{_libdir}/asterisk/modules/func_holdintercept.so %{_libdir}/asterisk/modules/func_iconv.so %{_libdir}/asterisk/modules/func_jitterbuffer.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_periodic_hook.so %{_libdir}/asterisk/modules/func_pitchshift.so %{_libdir}/asterisk/modules/func_presencestate.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_sorcery.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_talkdetect.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_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_agi.so %{_libdir}/asterisk/modules/res_ari.so %{_libdir}/asterisk/modules/res_ari_applications.so %{_libdir}/asterisk/modules/res_ari_asterisk.so %{_libdir}/asterisk/modules/res_ari_bridges.so %{_libdir}/asterisk/modules/res_ari_channels.so %{_libdir}/asterisk/modules/res_ari_device_states.so %{_libdir}/asterisk/modules/res_ari_endpoints.so %{_libdir}/asterisk/modules/res_ari_events.so #%{_libdir}/asterisk/modules/res_ari_mailboxes.so %{_libdir}/asterisk/modules/res_ari_model.so %{_libdir}/asterisk/modules/res_ari_playbacks.so %{_libdir}/asterisk/modules/res_ari_recordings.so %{_libdir}/asterisk/modules/res_ari_sounds.so %{_libdir}/asterisk/modules/res_chan_stats.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 %{_libdir}/asterisk/modules/res_endpoint_stats.so %{_libdir}/asterisk/modules/res_format_attr_celt.so %{_libdir}/asterisk/modules/res_format_attr_g729.so %{_libdir}/asterisk/modules/res_format_attr_h263.so %{_libdir}/asterisk/modules/res_format_attr_h264.so %{_libdir}/asterisk/modules/res_format_attr_ilbc.so %{_libdir}/asterisk/modules/res_format_attr_opus.so %{_libdir}/asterisk/modules/res_format_attr_silk.so %{_libdir}/asterisk/modules/res_format_attr_vp8.so %{_libdir}/asterisk/modules/res_format_attr_siren14.so %{_libdir}/asterisk/modules/res_format_attr_siren7.so %if (0%{?fedora} > 0 || 0%{?rhel} >= 7) && 0%{?gmime} %{_libdir}/asterisk/modules/res_http_post.so %endif %{_libdir}/asterisk/modules/res_http_media_cache.so %{_libdir}/asterisk/modules/res_http_websocket.so %{_libdir}/asterisk/modules/res_limit.so %{_libdir}/asterisk/modules/res_manager_devicestate.so %{_libdir}/asterisk/modules/res_manager_presencestate.so %{_libdir}/asterisk/modules/res_monitor.so %{_libdir}/asterisk/modules/res_musiconhold.so %{_libdir}/asterisk/modules/res_mutestream.so %{_libdir}/asterisk/modules/res_mwi_devstate.so %{_libdir}/asterisk/modules/res_parking.so %{_libdir}/asterisk/modules/res_phoneprov.so %{_libdir}/asterisk/modules/res_realtime.so # Now needed by res_rtp_asterisk otherwise there's a missing symbol %{_libdir}/asterisk/modules/res_pjproject.so %{_libdir}/asterisk/modules/res_prometheus.so %{_libdir}/asterisk/modules/res_remb_modifier.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_sorcery_astdb.so %{_libdir}/asterisk/modules/res_sorcery_config.so %{_libdir}/asterisk/modules/res_sorcery_memory.so %{_libdir}/asterisk/modules/res_sorcery_memory_cache.so %{_libdir}/asterisk/modules/res_sorcery_realtime.so %{_libdir}/asterisk/modules/res_speech.so %{_libdir}/asterisk/modules/res_srtp.so %{_libdir}/asterisk/modules/res_stasis.so %{_libdir}/asterisk/modules/res_stasis_answer.so %{_libdir}/asterisk/modules/res_stasis_device_state.so %{_libdir}/asterisk/modules/res_stasis_playback.so %{_libdir}/asterisk/modules/res_stasis_recording.so %{_libdir}/asterisk/modules/res_stasis_snoop.so %{_libdir}/asterisk/modules/res_statsd.so %{_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}/astcanary %{_sbindir}/astdb2sqlite3 %{_sbindir}/asterisk %{_sbindir}/astgenkey %{_sbindir}/astman %{_sbindir}/astversion %{_sbindir}/autosupport %{_sbindir}/check_expr %{_sbindir}/check_expr2 %{_sbindir}/muted %{_sbindir}/rasterisk #%{_sbindir}/refcounter %if ! %{systemd} %{_sbindir}/safe_asterisk %endif %{_sbindir}/smsq %{_sbindir}/stereorize %{_sbindir}/streamplayer %{_mandir}/man8/astdb2bdb.8* %{_mandir}/man8/astdb2sqlite3.8* %{_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/acl.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/adsi.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/ari.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/ast_debug_tools.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/confbridge.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.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/http.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/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/phoneprov.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/prometheus.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_parking.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_stun_monitor.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/resolver_unbound.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/sla.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/smdi.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/sorcery.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/stasis.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/statsd.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 %{_datadir}/asterisk/images %attr(0750,asterisk,asterisk) %{_datadir}/asterisk/keys %{_datadir}/asterisk/phoneprov %{_datadir}/asterisk/static-http %{_datadir}/asterisk/scripts %{_datadir}/asterisk/rest-api %dir %{_datadir}/asterisk/moh %dir %{_datadir}/asterisk/sounds %attr(0755,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 %endif %attr(0755,asterisk,asterisk) %dir %{astvarrundir} %files ael %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/extensions.ael %{_sbindir}/aelparse %{_sbindir}/conf2ael %{_libdir}/asterisk/modules/pbx_ael.so %{_libdir}/asterisk/modules/res_ael_share.so %files alsa %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/alsa.conf %{_libdir}/asterisk/modules/chan_alsa.so %if %{?apidoc} %files apidoc %defattr(-,root,root,-) %doc doc/api/html/* %endif %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 %if 0%{?xmpp} %{_libdir}/asterisk/modules/res_calendar_exchange.so %endif %{_libdir}/asterisk/modules/res_calendar_icalendar.so %if 0%{?corosync} %files corosync %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_corosync.conf %{_libdir}/asterisk/modules/res_corosync.so %endif %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 %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/ss7.timers %{_libdir}/asterisk/modules/app_flash.so %{_libdir}/asterisk/modules/app_meetme.so %{_libdir}/asterisk/modules/app_page.so %{_libdir}/asterisk/modules/app_dahdiras.so %{_libdir}/asterisk/modules/chan_dahdi.so %{_libdir}/asterisk/modules/codec_dahdi.so %{_libdir}/asterisk/modules/res_timing_dahdi.so %{_datadir}/dahdi/span_config.d/40-asterisk %files devel %defattr(-,root,root,-) %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 %files iax2 %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/iax.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/iaxprov.conf %dir %{_datadir}/asterisk/firmware %dir %{_datadir}/asterisk/firmware/iax %{_libdir}/asterisk/modules/chan_iax2.so %files hep %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/hep.conf %{_libdir}/asterisk/modules/res_hep.so %{_libdir}/asterisk/modules/res_hep_rtcp.so %{_libdir}/asterisk/modules/res_hep_pjsip.so %if 0%{?fedora} || 0%{?rhel} >= 7 %files ices %defattr(-,root,root,-) %doc contrib/asterisk-ices.xml %{_libdir}/asterisk/modules/app_ices.so %endif %if 0%{?jack} %files jack %defattr(-,root,root,-) %{_libdir}/asterisk/modules/app_jack.so %endif %files lua %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/extensions.lua %{_libdir}/asterisk/modules/pbx_lua.so %if 0%{?ldap} %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 %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 %if %{misdn} %files misdn %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/misdn.conf %{_libdir}/asterisk/modules/chan_misdn.so %endif %files mgcp %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/mgcp.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_pktccops.conf %{_libdir}/asterisk/modules/chan_mgcp.so %{_libdir}/asterisk/modules/res_pktccops.so %files mobile %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/chan_mobile.conf %{_libdir}/asterisk/modules/chan_mobile.so %files mp3 %defattr(-,root,root,-) %{_libdir}/asterisk/modules/format_mp3.so %if %{mysql} %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 %endif %files mwi-external %defattr(-,root,root,-) %{_libdir}/asterisk/modules/res_mwi_external.so %{_libdir}/asterisk/modules/res_mwi_external_ami.so %{_libdir}/asterisk/modules/res_stasis_mailbox.so %if %{odbc} %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 %{_libdir}/asterisk/modules/res_odbc_transaction.so %endif %files ooh323 %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/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 %if 0%{?xmpp} %files phone %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/phone.conf %{_libdir}/asterisk/modules/chan_phone.so %endif %files pjsip %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/pjproject.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/pjsip.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/pjsip_notify.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/pjsip_wizard.conf %{_libdir}/asterisk/modules/chan_pjsip.so %{_libdir}/asterisk/modules/func_pjsip_aor.so %{_libdir}/asterisk/modules/func_pjsip_contact.so %{_libdir}/asterisk/modules/func_pjsip_endpoint.so %{_libdir}/asterisk/modules/res_pjsip.so %{_libdir}/asterisk/modules/res_pjsip_acl.so %{_libdir}/asterisk/modules/res_pjsip_authenticator_digest.so %{_libdir}/asterisk/modules/res_pjsip_caller_id.so %{_libdir}/asterisk/modules/res_pjsip_config_wizard.so %{_libdir}/asterisk/modules/res_pjsip_dialog_info_body_generator.so %{_libdir}/asterisk/modules/res_pjsip_diversion.so %{_libdir}/asterisk/modules/res_pjsip_dlg_options.so %{_libdir}/asterisk/modules/res_pjsip_dtmf_info.so %{_libdir}/asterisk/modules/res_pjsip_empty_info.so %{_libdir}/asterisk/modules/res_pjsip_endpoint_identifier_anonymous.so %{_libdir}/asterisk/modules/res_pjsip_endpoint_identifier_ip.so %{_libdir}/asterisk/modules/res_pjsip_endpoint_identifier_user.so %{_libdir}/asterisk/modules/res_pjsip_exten_state.so %{_libdir}/asterisk/modules/res_pjsip_header_funcs.so %{_libdir}/asterisk/modules/res_pjsip_history.so #%{_libdir}/asterisk/modules/res_pjsip_keepalive.so %{_libdir}/asterisk/modules/res_pjsip_logger.so %{_libdir}/asterisk/modules/res_pjsip_messaging.so #%{_libdir}/asterisk/modules/res_pjsip_multihomed.so %{_libdir}/asterisk/modules/res_pjsip_mwi.so %{_libdir}/asterisk/modules/res_pjsip_mwi_body_generator.so %{_libdir}/asterisk/modules/res_pjsip_nat.so %{_libdir}/asterisk/modules/res_pjsip_notify.so %{_libdir}/asterisk/modules/res_pjsip_one_touch_record_info.so %{_libdir}/asterisk/modules/res_pjsip_outbound_authenticator_digest.so %{_libdir}/asterisk/modules/res_pjsip_outbound_publish.so %{_libdir}/asterisk/modules/res_pjsip_outbound_registration.so %{_libdir}/asterisk/modules/res_pjsip_path.so %{_libdir}/asterisk/modules/res_pjsip_phoneprov_provider.so %{_libdir}/asterisk/modules/res_pjsip_pidf_body_generator.so %{_libdir}/asterisk/modules/res_pjsip_pidf_digium_body_supplement.so %{_libdir}/asterisk/modules/res_pjsip_pidf_eyebeam_body_supplement.so %{_libdir}/asterisk/modules/res_pjsip_publish_asterisk.so %{_libdir}/asterisk/modules/res_pjsip_pubsub.so %{_libdir}/asterisk/modules/res_pjsip_refer.so %{_libdir}/asterisk/modules/res_pjsip_registrar.so #%{_libdir}/asterisk/modules/res_pjsip_registrar_expire.so %{_libdir}/asterisk/modules/res_pjsip_rfc3326.so %{_libdir}/asterisk/modules/res_pjsip_sdp_rtp.so %{_libdir}/asterisk/modules/res_pjsip_send_to_voicemail.so %{_libdir}/asterisk/modules/res_pjsip_session.so %{_libdir}/asterisk/modules/res_pjsip_sips_contact.so %{_libdir}/asterisk/modules/res_pjsip_t38.so #%{_libdir}/asterisk/modules/res_pjsip_transport_management.so %{_libdir}/asterisk/modules/res_pjsip_transport_websocket.so %{_libdir}/asterisk/modules/res_pjsip_xpidf_body_generator.so #%{_libdir}/asterisk/modules/res_sdp_translator_pjmedia.so %files portaudio %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/console.conf %{_libdir}/asterisk/modules/chan_console.so %if %{postgresql} %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 %endif %if %{radius} %files radius %defattr(-,root,root,-) %{_libdir}/asterisk/modules/cdr_radius.so %{_libdir}/asterisk/modules/cel_radius.so %endif %files sip %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/sip.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/sip_notify.conf %{_libdir}/asterisk/modules/chan_sip.so %files skinny %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/skinny.conf %{_libdir}/asterisk/modules/chan_skinny.so %if %{snmp} %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 %endif %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 %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/res_config_sqlite3.conf %{_libdir}/asterisk/modules/cdr_sqlite3_custom.so %{_libdir}/asterisk/modules/cel_sqlite3_custom.so %{_libdir}/asterisk/modules/res_config_sqlite3.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,-) %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 %{_libdir}/asterisk/modules/app_directory.so %{_libdir}/asterisk/modules/app_voicemail.so %if 0%{?fedora} > 0 || 0%{?rhel} >= 7 %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 %if 0%{?xmpp} %files xmpp %defattr(-,root,root,-) %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/motif.conf %attr(0640,asterisk,asterisk) %config(noreplace) %{_sysconfdir}/asterisk/xmpp.conf %{_libdir}/asterisk/modules/chan_motif.so %{_libdir}/asterisk/modules/res_xmpp.so %endif %changelog * Thu Feb 25 2021 Ganapathi Chidambaram <ganapathi.chidambaram@flexydial.com> - 17.9.2-1 - New version * Thu Apr 9 2015 Jeffrey C. Ollie <jeff@ocjtech.us> - 13.3.2-1: - The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available - security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11, - 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following security vulnerability: - - * AST-2015-003: TLS Certificate Common name NULL byte exploit - - When Asterisk registers to a SIP TLS device and and verifies the server, - Asterisk will accept signed certificates that match a common name other than - the one Asterisk is expecting if the signed certificate has a common name - containing a null byte after the portion of the common name that Asterisk - expected. This potentially allows for a man in the middle attack. - - For more information about the details of this vulnerability, please read - security advisory AST-2015-003, which was released at the same time as this - announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2 - - The security advisory is available at: - - * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf * Thu Apr 9 2015 Jeffrey C. Ollie <jeff@ocjtech.us> - 13.3.1-1: - The Asterisk Development Team has announced the release of Asterisk 13.3.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 13.3.1 resolves an issue reported by the - community and would have not been possible without your participation. - Thank you! - - The following is the issue resolved in this release: - - * --- pjsip: resolve compatibility problem with ast_sip_session - (Closes issue ASTERISK-24941. Reported by Matt Jordan) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.1 * Wed Apr 1 2015 Jeffrey C. Ollie <jeff@ocjtech.us> - 13.3.0-1: - The Asterisk Development Team has announced the release of Asterisk 13.3.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 13.3.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - New Features made in this release: - ----------------------------------- - * ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a - channel (Reported by Matt Jordan) - * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation - (Reported by Dwayne Hubbard) - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid - string copy (Reported by Yura Kocyuba) - * ASTERISK-24748 - res_pjsip: If wizards explicitly configured in - sorcery.conf false ERROR messages may occur (Reported by Joshua - Colp) - * ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked - (Reported by Matt Jordan) - * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in - res_odbc (Reported by ibercom) - * ASTERISK-24479 - Enable REF_DEBUG for module references - (Reported by Corey Farrell) - * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to - fully disconnect underlying socket, leading to events being - dropped with no additional information (Reported by Matt Jordan) - * ASTERISK-24772 - ODBC error in realtime sippeers when device - unregisters under MariaDB (Reported by Richard Miller) - * ASTERISK-24752 - Crash in bridge_manager_service_req when bridge - is destroyed by ARI during shutdown (Reported by Richard - Mudgett) - * ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported - by Zane Conkle) - * ASTERISK-24015 - app_transfer fails with PJSIP channels - (Reported by Private Name) - * ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk - transfer scenario. (Reported by Mark Michelson) - * ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by - Niklas Larsson) - * ASTERISK-24716 - Improve pjsip log messages for presence - subscription failure (Reported by Rusty Newton) - * ASTERISK-24612 - res_pjsip: No information if a required sorcery - wizard is not loaded (Reported by Joshua Colp) - * ASTERISK-24768 - res_timing_pthread: file descriptor leak - (Reported by Matthias Urlichs) - * ASTERISK-24685 - "pjsip show version" CLI command (Reported by - Joshua Colp) - * ASTERISK-24632 - install_prereq script installs pjproject - without IPv6 support (Reported by Rusty Newton) - * ASTERISK-24085 - Documentation - We should remove or further - document the 'contact' section in pjsip.conf (Reported by Rusty - Newton) - * ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by - JoshE) - * ASTERISK-24700 - CRASH: NULL channel is being passed to - ast_bridge_transfer_attended() (Reported by Zane Conkle) - * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove - (Reported by Corey Farrell) - * ASTERISK-24799 - [patch] make fails with undefined reference to - SSLv3_client_method (Reported by Alexander Traud) - * ASTERISK-22670 - Asterisk crashes when processing ISDN AoC - Events (Reported by klaus3000) - * ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn - call (Reported by Marcel Manz) - * ASTERISK-24740 - [patch]Segmentation fault on aoc-e event - (Reported by Panos Gkikakis) - * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility - for playing back messages stored in IMAP - play_message: No - origtime (Reported by Graham Barnett) - * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc - OSX with 64 bit integers (Reported by Corey Farrell) - * ASTERISK-24796 - Codecs and bucket schema's prevent module - unload (Reported by Corey Farrell) - * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML - (Reported by Ashley Sanders) - * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring - is invalid (Reported by Rusty Newton) - * ASTERISK-24785 - 'Expires' header missing from 200 OK on - REGISTER (Reported by Ross Beer) - * ASTERISK-24677 - ARI GET variable on channel provides unhelpful - response on non-existent variable (Reported by Joshua Colp) - * ASTERISK-24797 - bridge_softmix: G.729 codec license held - (Reported by Kevin Harwell) - * ASTERISK-24812 - ARI: Creating channels through /channels - resource always uses SLIN, which results in unneeded transcoding - (Reported by Matt Jordan) - * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid - thread ID being passed to pthread_kill (Reported by JoshE) - * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime - fail (Reported by Terry Wilson) - * ASTERISK-23214 - chan_sip WARNING message 'We are requesting - SRTP for audio, but they responded without it' is ambiguous and - wrong in some cases (Reported by Rusty Newton) - * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an - error response and BYE are sent to the caller (Reported by - Makoto Dei) - * ASTERISK-18105 - most of asterisk modules are unbuildable in - cygwin environment (Reported by feyfre) - * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell) - * ASTERISK-24751 - Integer values in json payload to ARI cause - asterisk to crash (Reported by jeffrey putnam) - * ASTERISK-24838 - chan_sip: Locking inversion occurs when - building a peer causes a peer poke during request handling - (Reported by Richard Mudgett) - * ASTERISK-24825 - Caller ID not recognized using - Centrex/Distinctive dialing (Reported by Richard Mudgett) - * ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not - HAVE_PJPROJECT (Reported by Stefan Engström) - * ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers - (Reported by Kevin Harwell) - * ASTERISK-24755 - Asterisk sends unexpected early BYE to - transferrer during attended transfer when using a Stasis bridge - (Reported by John Bigelow) - * ASTERISK-24739 - [patch] - Out of files -- call fails -- - numerous files with inodes from under /usr/share/zoneinfo, - mostly posixrules (Reported by Ed Hynan) - * ASTERISK-23390 - NewExten Event with application AGI shows up - before and after AGI runs (Reported by Benjamin Keith Ford) - * ASTERISK-24786 - [patch] - Asterisk terminates when playing a - voicemail stored in LDAP (Reported by Graham Barnett) - * ASTERISK-24808 - res_config_odbc: Improper escaping of - backslashes occurs with MySQL (Reported by Javier Acosta) - * ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported - by Anatoli) - * ASTERISK-20850 - [patch]Nested functions aren't portable. - Adapting RAII_VAR to use clang/llvm blocks to get the - same/similar functionality. (Reported by Diederik de Groot) - * ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI - connection on error (Reported by Dmitriy Serov) - * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported - by Frank DiGennaro) - * ASTERISK-21038 - Bad command completion of "core set debug - channel" (Reported by Richard Kenner) - * ASTERISK-18708 - func_curl hangs channel under load (Reported by - Dave Cabot) - * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by - Atis Lezdins) - * ASTERISK-24876 - Investigate reference leaks from - tests/channels/local/local_optimize_away (Reported by Corey - Farrell) - * ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported - by Corey Farrell) - * ASTERISK-24817 - init_logger_chain: unreachable code block - (Reported by Corey Farrell) - * ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by - snuffy) - * ASTERISK-24879 - [patch]Compilation fails due to 64bit time - under OpenBSD (Reported by snuffy) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes - (Reported by Ben Merrills) - * ASTERISK-24811 - asterisk-publication sorcery object does not - use realtime (Reported by Matt Hoskins) - * ASTERISK-24790 - Reduce spurious noise in logs from voicemail - - Couldn't find mailbox %s in context (Reported by Graham Barnett) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0 * Wed Apr 1 2015 Jeffrey C. Ollie <jeff@ocjtech.us> - 13.2.0-1: - The Asterisk Development Team has announced the release of Asterisk 13.2.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 13.2.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-24342 - PJSIP: Qualifying endpoints attempts to do them - all at the same time. (Reported by Richard Mudgett) - * ASTERISK-24514 - res_pjsip_outbound_registration: stack overflow - when using non-default sorcery wizard (Reported by Kevin - Harwell) - * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS - from JSSIP (Reported by Badalian Vyacheslav) - * ASTERISK-24607 - res_pjsip_session: re-INVITE with declined - media streams results in 488 (Reported by Matt Jordan) - * ASTERISK-24563 - Direct Media calls within private network - sometimes get one way audio (Reported by Kevin Harwell) - * ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to - race condition in accessing codec in stored ast_frame and codec - core (Reported by Matt Jordan) - * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag - enabled (Reported by Richard Mudgett) - * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is - enabled (Reported by Andreas Steinmetz) - * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly - casts char to unsigned int (Reported by Walter Doekes) - * ASTERISK-24536 - AMI redirect with PJSIP fails to move extra - channel (Reported by Niklas Larsson) - * ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is - chosen for RTP compatible channels when the DTMF mode is not - compatible (Reported by Yaniv Simhi) - * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher - level - 'Remote address is null, most likely RTP has been - stopped' (Reported by Rusty Newton) - * ASTERISK-24513 - Local channel apparently leaked in off-nominal - DTMF attended transfer (Reported by Mark Michelson) - * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present - on startup (Reported by Richard Kenner) - * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong - destination when 'sendrpid=yes' (in proxy environment) (Reported - by Karsten Wemheuer) - * ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall - calls to the transferrer. (Reported by Richard Mudgett) - * ASTERISK-24376 - res_pjsip_refer: REFER request for remote - session attempts to direct channel to external_replaces - extension instead of context, without providing for the - Referred-To SIP URI (Reported by Matt Jordan) - * ASTERISK-24591 - Stasis() side of an ARI originated channel - cannot be Redirected (Reported by Kinsey Moore) - * ASTERISK-24049 - Asterisk Manager Interface: A number of list - type responses aren't using astman_send_listack (Reported by - Jonathan Rose) - * ASTERISK-24637 - Channel re-enters Stasis() when it should not - (Reported by John Bigelow) - * ASTERISK-24474 - sip_to_pjsip.py lacks documentation and does - not function (Reported by John Kiniston) - * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT - (Reported by Kristian Høgh) - * ASTERISK-20744 - [patch] Security event logging does not work - over syslog (Reported by Michael Keuter) - * ASTERISK-24665 - Configure check required for - pjsip_get_dest_info() (Reported by Mark Michelson) - * ASTERISK-23850 - Park Application does not respect Return - Context Priority (Reported by Andrew Nagy) - * ASTERISK-23991 - [patch]asterisk.pc file contains a small error - in the CFlags returned (Reported by Diederik de Groot) - * ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdown - while attempting to publish (Reported by Kevin Harwell) - * ASTERISK-24485 - res_pjsip cannot be unloaded or shutdown - (Reported by Corey Farrell) - * ASTERISK-24663 - [patch] Unnamed semaphore autoconf check fails - on cross compilation (Reported by abelbeck) - * ASTERISK-24624 - Transfer to invalid extension results in hung - channel. (Reported by Zane Conkle) - * ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf, - Incorrect External Addresses is Used in SIP Packets When - Responding to INVITE (Reported by David Justl) - * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail - - voicemail is not deleted after review, hangup (Reported by LEI - FU) - * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects - 32-bit packages on 64-bit hosts (Reported by Ben Klang) - * ASTERISK-24600 - Stuck IAX channels, Asterisk stops responding - to most traffic, potential deadlock (Reported by Jeff Collell) - * ASTERISK-24560 - Creating a named ARI bridge twice causes a - crash (Reported by Kinsey Moore) - * ASTERISK-24682 - app_dial: Multiple DialEnd events emitted when - MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported - by Matt Jordan) - * ASTERISK-24640 - Registration pending stays forever after sip - reload (Reported by Max Man) - * ASTERISK-24673 - outgoing sip registers cannot be removed or - modified without doing restart (or doing module unload - chan_sip.so) (Reported by Stefan Engström) - * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor - m() option does not queue an MWI event (Reported by Gareth - Palmer) - * ASTERISK-24649 - Pushing of channel into bridge fails; Stasis - fails to get app name (Reported by John Bigelow) - * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive - column comparison for 'defaultuser' (Reported by - HZMI8gkCvPpom0tM) - * ASTERISK-24693 - Investigate and fix memory leaks in Asterisk - (Reported by Kevin Harwell) - * ASTERISK-24626 - Voicemail passwords not being stored in ARA - (Reported by Paddy Grice) - * ASTERISK-24539 - Compile fails on OSX because of sem_timedwait - in bridge_channel.c (Reported by George Joseph) - * ASTERISK-24544 - Compile fails on OSX Yosemite because of - incorrect detection of htonll and ntohll (Reported by George - Joseph) - * ASTERISK-24723 - confbridge: CLI command 'confbridge list XXXX' - no longer displays user menus (Reported by Matt Jordan) - * ASTERISK-24721 - manager: ModuleLoad action incorrectly reports - 'module not found' during a Reload operation (Reported by Matt - Jordan) - * ASTERISK-24719 - ConfBridge recording channels get stuck when - recording started/stopped more than once (Reported by Richard - Mudgett) - * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported - by Kevin Harwell) - * ASTERISK-24728 - tcptls: Bad file descriptor error when - reloading chan_sip (Reported by Kevin Harwell) - * ASTERISK-24729 - Outbound registration not occuring on new - registrations after reload. (Reported by Richard Mudgett) - * ASTERISK-24676 - Security Vulnerability: URL request injection - in libCURL (CVE-2014-8150) (Reported by Matt Jordan) - * ASTERISK-24666 - Security Vulnerability: RTP not closed after - sip call using unsupported codec (Reported by Y Ateya) - * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL - versions (Reported by Jared Biel) - * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by - Stephan Eisvogel) - * ASTERISK-24736 - Memory Leak Fixes (Reported by Mark Michelson) - * ASTERISK-24635 - PJSIP outbound PUBLISH crashes when no response - is ever received (Reported by Marco Paland) - * ASTERISK-24737 - When agent not logged in, agent status shows - unavailable, queue status shows agent invalid (Reported by - Richard Mudgett) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-24552 - ARI: Allow associating a channel as an - initiator of an Origination for record keeping purposes - (Reported by Matt Jordan) - * ASTERISK-24553 - ARI/AMI: Include language in standard channel - snapshot output (Reported by Matt Jordan) - * ASTERISK-24643 - res_pjsip: Add user=phone option (Reported by - Matt Jordan) - * ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for - connection-oriented transports. (Reported by Matt Jordan) - * ASTERISK-24412 - [patch]Incomplete channel originate/continue - handling with ARI (Reported by Nir Simionovich (GreenfieldTech - - Israel)) - * ASTERISK-24678 - [PATCH] Added atxfer* settings to - features.conf.sample (Reported by Niklas Larsson) - * ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported - by cloos) - * ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by - Dan Jenkins) - * ASTERISK-24316 - For httpd server, need option to define server - name for security purposes (Reported by Andrew Nagy) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0 * Fri Jan 30 2015 Jeffrey C. Ollie <jeff@ocjtech.us> - 13.1.1-1: - The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available - security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10, - 11.15.1, 12.8.1, and 13.1.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following security vulnerabilities: - - * AST-2015-001: File descriptor leak when incompatible codecs are offered - - Asterisk may be configured to only allow specific audio or - video codecs to be used when communicating with a - particular endpoint. When an endpoint sends an SDP offer - that only lists codecs not allowed by Asterisk, the offer - is rejected. However, in this case, RTP ports that are - allocated in the process are not reclaimed. - - This issue only affects the PJSIP channel driver in - Asterisk. Users of the chan_sip channel driver are not - affected. - - * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability - - CVE-2014-8150 reported an HTTP request injection - vulnerability in libcURL. Asterisk uses libcURL in its - func_curl.so module (the CURL() dialplan function), as well - as its res_config_curl.so (cURL realtime backend) modules. - - Since Asterisk may be configured to allow for user-supplied - URLs to be passed to libcURL, it is possible that an - attacker could use Asterisk as an attack vector to inject - unauthorized HTTP requests if the version of libcURL - installed on the Asterisk server is affected by - CVE-2014-8150. - - For more information about the details of these vulnerabilities, please read - security advisory AST-2015-001 and AST-2015-002, which were released at the same - time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf - * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf * Fri Jan 30 2015 Jeffrey C. Ollie <jeff@ocjtech.us> - 13.1.0-1 - The Asterisk Development Team has announced the release of Asterisk 13.1.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 13.1.0 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: - - New Features made in this release: - ----------------------------------- - * ASTERISK-24554 - AMI/ARI: Generate events on connected line - changes (Reported by Matt Jordan) - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling - against libsrtp-1.5.0 (Reported by Patrick Laimbock) - * ASTERISK-24455 - func_cdr: CDR_PROP leaks payload (Reported by - Corey Farrell) - * ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing - leak (Reported by Corey Farrell) - * ASTERISK-24430 - missing letter "p" in word response in - OriginateResponse event documentation (Reported by Dafi Ni) - * ASTERISK-24437 - Review implementation of ast_bridge_impart for - leaks and document proper usage (Reported by Scott Griepentrog) - * ASTERISK-24453 - manager: acl_change_sub leaks (Reported by - Corey Farrell) - * ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by - Corey Farrell) - * ASTERISK-24458 - chan_phone fails to build on big endian systems - (Reported by Tzafrir Cohen) - * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers - (Reported by Olle Johansson) - * ASTERISK-24304 - asterisk crashing randomly because of unistim - channel (Reported by dhanapathy sathya) - * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by - Nick Adams) - * ASTERISK-24462 - res_pjsip: Stale qualify statistics after - disablementation (Reported by Kevin Harwell) - * ASTERISK-24465 - audiohooks list leaks reference to formats - (Reported by Corey Farrell) - * ASTERISK-24466 - app_queue: fix a couple leaks to struct - call_queue (Reported by Corey Farrell) - * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled - (Reported by Corey Farrell) - * ASTERISK-24411 - [patch] Status of outbound registration is not - changed upon unregistering. (Reported by John Bigelow) - * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream - leaks (Reported by Corey Farrell) - * ASTERISK-24480 - res_http_websockets: Module reference decrease - below zero (Reported by Corey Farrell) - * ASTERISK-24482 - func_talkdetect: Fix stasis message leak in - audiohook callback (Reported by Corey Farrell) - * ASTERISK-24487 - configuration: sections should be loadable as - template even when not marked (Reported by Scott Griepentrog) - * ASTERISK-20127 - [Regression] Config.c config_text_file_load() - unescapes semicolons ("\;" -> ";") turning them into comments - (corruption) on rewrite of a config file (Reported by George - Joseph) - * ASTERISK-24438 - res_pjsip_multihomed.so blocks Asterisk reload - when DNS settings invalid (Reported by Melissa Shepherd) - * ASTERISK-24307 - Unintentional memory retention in stringfields - (Reported by Etienne Lessard) - * ASTERISK-24491 - Memory leak in res_hep (Reported by Zane - Conkle) - * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes - extra calls to ast_module_unref (Reported by Corey Farrell) - * ASTERISK-24447 - Bridge DTMF hooks: Audio doesn't pass when - waiting for more matching digits. (Reported by Richard Mudgett) - * ASTERISK-24257 - agent must dial acceptdtmf twice to bridge to - queue caller (Reported by Steve Pitts) - * ASTERISK-24504 - chan_console: Fix reference leaks to pvt - (Reported by Corey Farrell) - * ASTERISK-24250 - [patch] Voicemail with multi-recipients To: - header fix (Reported by abelbeck) - * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS - length exceeds 50 (roughly) national symbols (Reported by - Dmitriy Bubnov) - * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN - revision r227276 (Reported by Xavier Hienne) - * ASTERISK-24505 - manager: http connections leak references - (Reported by Corey Farrell) - * ASTERISK-24502 - Build fails when dev-mode, dont optimize and - coverage are enabled (Reported by Corey Farrell) - * ASTERISK-24444 - PBX: Crash when generating extension for - pattern matching hint (Reported by Leandro Dardini) - * ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP - packet to JSON for res_hep_rtcp and report blocks are greater - than 1 (Reported by Gregory Malsack) - * ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended - transfer (Reported by Beppo Mazzucato) - * ASTERISK-24501 - ARI: Moving a channel between bridges followed - by a hangup can cause an ARI client to not receive an expected - ChannelLeftBridge event before StasisEnd (Reported by Matt - Jordan) - * ASTERISK-24336 - PJSIP timer_min_se value under 90 causes crash - (Reported by Leon Rowland) - * ASTERISK-23651 - Reloading some modules that are loaded already, - results in 'No such module' before a successful reload (Reported - by Rusty Newton) - * ASTERISK-24522 - ConfBridge: delay occurs between kicking all - endmarked users when last marked user leaves (Reported by Matt - Jordan) - * ASTERISK-15242 - transmit_refer leaks sip_refer structures - (Reported by David Woolley) - * ASTERISK-24508 - pjsip - REFER request from SNOM is rejected - with "400 bad request" - DEBUG shows "Received a REFER without a - parseable Refer-To" (Reported by Beppo Mazzucato) - * ASTERISK-24535 - stringfields: Fix regression from fix for - unintentional memory retention and another issue exposed by the - fix (Reported by Corey Farrell) - * ASTERISK-24471 - Crash - assert_fail in libc in - pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2 - (Reported by yaron nahum) - * ASTERISK-24528 - res_pjsip_refer: Sending INVITE with Replaces - in-dialog with invalid target causes crash (Reported by Joshua - Colp) - * ASTERISK-24531 - res_pjsip_acl: ACLs not applied on initial - module load (Reported by Matt Jordan) - * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs - allow blocked addresses through (Reported by Matt Jordan) - * ASTERISK-24542 - [patch]Failure showing codecs via 'core show - channeltype <tech>' (Reported by snuffy) - * ASTERISK-24533 - 2 threads created per chan_sip entry (Reported - by xrobau) - * ASTERISK-24516 - [patch]Asterisk segfaults when playing back - voicemail under high concurrency with an IMAP backend (Reported - by David Duncan Ross Palmer) - * ASTERISK-24572 - [patch]App_meetme is loaded without its - defaults when the configuration file is missing (Reported by - Nuno Borges) - * ASTERISK-24573 - [patch]Out of sync conversation recording when - divided in multiple recordings (Reported by Nuno Borges) - * ASTERISK-24537 - Stasis: StasisStart/StasisEnd events are not - reliably transmitted during transfers (Reported by Matt Jordan) - * ASTERISK-24556 - Asterisk 13 core dumps when calling from pjsip - extension to another pjsip extension (Reported by Abhay Gupta) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR - property 'unanswered' (Reported by Matt Jordan) - * ASTERISK-24283 - [patch]Microseconds precision in the eventtime - column in the cel_odbc module (Reported by Etienne Lessard) - * ASTERISK-24530 - [patch] app_record stripping 1/4 second from - recordings (Reported by Ben Smithurst) - * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded - lookups (Reported by Birger "WIMPy" Harzenetter) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.1.0 * Thu Jan 29 2015 Peter Robinson <pbrobinson@fedoraproject.org> 13.0.2-3 - Add speexdsp as build dep as speex_echo.h has moved - rhbz 1181021 * Thu Jan 15 2015 Tom Callaway <spot@fedoraproject.org> - 13.0.2-2 - update for lua 5.3 * Wed Dec 10 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 13.0.2-1 - The Asterisk Development Team has announced security releases for Certified - Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are - released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following security vulnerability: - - * AST-2014-019: Remote Crash Vulnerability in WebSocket Server - - When handling a WebSocket frame the res_http_websocket module dynamically - changes the size of the memory used to allow the provided payload to fit. If a - payload length of zero was received the code would incorrectly attempt to - resize to zero. This operation would succeed and end up freeing the memory but - be treated as a failure. When the session was subsequently torn down this - memory would get freed yet again causing a crash. - - For more information about the details of this vulnerability, please read - security advisory AST-2014-019, which was 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/certified-asterisk/releases/ChangeLog-11.6-cert9 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2 - - The security advisory is available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf * Thu Nov 20 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 13.0.1-1 - The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available - security releases are released as versions 1.8.28-cert3, 11.6-cert8, 1.8.32.1, - 11.14.1, 12.7.1, and 13.0.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following security vulnerabilities: - - * AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP - address families - - Many modules in Asterisk that service incoming IP traffic have ACL options - ("permit" and "deny") that can be used to whitelist or blacklist address - ranges. A bug has been discovered where the address family of incoming - packets is only compared to the IP address family of the first entry in the - list of access control rules. If the source IP address for an incoming - packet is not of the same address as the first ACL entry, that packet - bypasses all ACL rules. - - * AST-2014-018: Permission Escalation through DB dialplan function - - The DB dialplan function when executed from an external protocol, such as AMI, - could result in a privilege escalation. Users with a lower class authorization - in AMI can access the internal Asterisk database without the required SYSTEM - class authorization. - - In addition, the release of 11.6-cert8 and 11.14.1 resolves the following - security vulnerability: - - * AST-2014-014: High call load with ConfBridge can result in resource exhaustion - - The ConfBridge application uses an internal bridging API to implement - conference bridges. This internal API uses a state model for channels within - the conference bridge and transitions between states as different things - occur. Unload load it is possible for some state transitions to be delayed - causing the channel to transition from being hung up to waiting for media. As - the channel has been hung up remotely no further media will arrive and the - channel will stay within ConfBridge indefinitely. - - In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves - the following security vulnerability: - - * AST-2014-017: Permission Escalation via ConfBridge dialplan function and - AMI ConfbridgeStartRecord Action - - The CONFBRIDGE dialplan function when executed from an external protocol (such - as AMI) can result in a privilege escalation as certain options within that - function can affect the underlying system. Additionally, the AMI - ConfbridgeStartRecord action has options that would allow modification of the - underlying system, and does not require SYSTEM class authorization in AMI. - - Finally, the release of 12.7.1 and 13.0.1 resolves the following security - vulnerabilities: - - * AST-2014-013: Unauthorized access in the presence of ACLs in the PJSIP stack - - The Asterisk module res_pjsip provides the ability to configure ACLs that may - be used to reject SIP requests from various hosts. However, the module - currently fails to create and apply the ACLs defined in its configuration - file on initial module load. - - * AST-2014-015: Remote crash vulnerability in PJSIP channel driver - - The chan_pjsip channel driver uses a queue approach for relating to SIP - sessions. There exists a race condition where actions may be queued to answer - a session or send ringing after a SIP session has been terminated using a - CANCEL request. The code will incorrectly assume that the SIP session is still - active and attempt to send the SIP response. The PJSIP library does not - expect the SIP session to be in the disconnected state when sending the - response and asserts. - - * AST-2014-016: Remote crash vulnerability in PJSIP channel driver - - When handling an INVITE with Replaces message the res_pjsip_refer module - incorrectly assumes that it will be operating on a channel that has just been - created. If the INVITE with Replaces message is sent in-dialog after a session - has been established this assumption will be incorrect. The res_pjsip_refer - module will then hang up a channel that is actually owned by another thread. - When this other thread attempts to use the just hung up channel it will end up - using a freed channel which will likely result in a crash. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015, - AST-2014-016, AST-2014-017, and AST-2014-018, which were released at the same - time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert3 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert8 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-013.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-014.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-015.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-016.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-017.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf * Thu Nov 20 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 13.0.0-1 - The Asterisk Development Team is pleased to announce the release of - Asterisk 13.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - Asterisk 13 is the next major release series of Asterisk. It is a Long Term - Support (LTS) release, similar to Asterisk 11. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - For important information regarding upgrading to Asterisk 13, please see the - Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13 - - A short list of new features includes: - - * Asterisk security events are now provided via AMI, allowing end users to - monitor their Asterisk system in real time for security related issues. - - * Both AMI and ARI now allow external systems to control the state of a mailbox. - Using AMI actions or ARI resources, external systems can programmatically - trigger Message Waiting Indicators (MWI) on subscribed phones. This is of - particular use to those who want to build their own VoiceMail application - using ARI. - - * ARI now supports the reception/transmission of out of call text messages using - any supported channel driver/protocol stack through ARI. Users receive out of - call text messages as JSON events over the ARI websocket connection, and can - send out of call text messages using HTTP requests. - - * The PJSIP stack now supports RFC 4662 Resource Lists, allowing Asterisk to act - as a Resource List Server. This includes defining lists of presence state, - mailbox state, or lists of presence state/mailbox state; managing - subscriptions to lists; and batched delivery of NOTIFY requests to - subscribers. - - * The PJSIP stack can now be used as a means of distributing device state or - mailbox state via PUBLISH requests to other Asterisk instances. This is - analogous to Asterisk's clustering support using XMPP or Corosync; unlike - existing clustering mechanisms, using the PJSIP stack to perform the - distribution of state does not rely on another daemon or server to perform the - work. - - And much more! - - More information about the new features can be found on the Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Documentation - - A full list of all new features can also be found in the CHANGES file: - - http://svnview.digium.com/svn/asterisk/branches/13/CHANGES - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0 * Fri Nov 14 2014 Tom Callaway <spot@fedoraproject.org> - 11.13.1-2 - rebuild for new libsrtp * Mon Oct 20 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 11.13.1-1 - The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available - security releases are released as versions 1.8.28-cert2, 11.6-cert7, 1.8.31.1, - 11.13.1, 12.6.1, and 13.0.0-beta3. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following security vulnerability: - - * AST-2014-011: Asterisk Susceptibility to POODLE Vulnerability - - Asterisk is susceptible to the POODLE vulnerability in two ways: - 1) The res_jabber and res_xmpp module both use SSLv3 exclusively for their - encrypted connections. - 2) The core TLS handling in Asterisk, which is used by the chan_sip channel - driver, Asterisk Manager Interface (AMI), and Asterisk HTTP Server, by - default allow a TLS connection to fallback to SSLv3. This allows for a - MITM to potentially force a connection to fallback to SSLv3, exposing it - to the POODLE vulnerability. - - These issues have been resolved in the versions released in conjunction with - this security advisory. - - For more information about the details of this vulnerability, please read - security advisory AST-2014-011, which was 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/certified-asterisk/releases/ChangeLog-1.8.28-cert2 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert7 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.31.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.13.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.6.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0-beta3 - - The security advisory is available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-011.pdf * Mon Oct 20 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 11.13.0-1 - The Asterisk Development Team has announced the release of Asterisk 11.13.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.13.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-24032 - Gentoo compilation emits warning: - "_FORTIFY_SOURCE" redefined (Reported by Kilburn) - * ASTERISK-24225 - Dial option z is broken (Reported by - dimitripietro) - * ASTERISK-24178 - [patch]fromdomainport used even if not set - (Reported by Elazar Broad) - * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload - warnings and ref leaks (Reported by Walter Doekes) - * ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP - ICE candidates in SDP answer (Reported by Badalian Vyacheslav) - * ASTERISK-24019 - When a Music On Hold stream starts it restarts - at beginning of file. (Reported by Jason Richards) - * ASTERISK-23767 - [patch] Dynamic IAX2 registration stops trying - if ever not able to resolve (Reported by David Herselman) - * ASTERISK-24211 - testsuite: Fix the dial_LS_options test - (Reported by Matt Jordan) - * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash - Mohod) - * ASTERISK-23577 - res_rtp_asterisk: Crash in - ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by - Jay Jideliov) - * ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10) - concurrent WebRTC (avpg/encryption/icesupport) calls (Reported - by Roman Skvirsky) - * ASTERISK-24301 - Security: Out of call MESSAGE requests - processed via Message channel driver can crash Asterisk - (Reported by Matt Jordan) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-24171 - [patch] Provide a manpage for the aelparse - utility (Reported by Jeremy Lainé) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.13.0 * Mon Oct 20 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 11.12.1-1 - The Asterisk Development Team has announced security releases for Certified - Asterisk 11.6 and Asterisk 11 and 12. The available security releases are - released as versions 11.6-cert6, 11.12.1, and 12.5.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - Please note that the release of these versions resolves the following security - vulnerability: - - * AST-2014-010: Remote Crash when Handling Out of Call Message in Certain - Dialplan Configurations - - Additionally, the release of Asterisk 12.5.1 resolves the following security - vulnerability: - - * AST-2014-009: Remote Crash Based on Malformed SIP Subscription Requests - - Note that the crash described in AST-2014-010 can be worked around through - dialplan configuration. Given the likelihood of the issue, an advisory was - deemed to be warranted. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2014-009 and AST-2014-010, 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/certified-asterisk/releases/ChangeLog-11.6-cert6 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.12.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.5.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-009.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-010.pdf * Mon Oct 20 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 11.12.0-1 - The Asterisk Development Team has announced the release of Asterisk 11.12.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.12.0 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: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-23911 - URIENCODE/URIDECODE: WARNING about passing an - empty string is a bit over zealous (Reported by Matt Jordan) - * ASTERISK-23985 - PresenceState Action response does not contain - ActionID; duplicates Message Header (Reported by Matt Jordan) - * ASTERISK-23814 - No call started after peer dialed (Reported by - Igor Goncharovsky) - * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy - should not call sip_destroy (Reported by Corey Farrell) - * ASTERISK-23818 - PBX_Lua: after asterisk startup module is - loaded, but dialplan not available (Reported by Dennis Guse) - * ASTERISK-18345 - [patch] sips connection dropped by asterisk - with a large INVITE (Reported by Stephane Chazelas) - * ASTERISK-23508 - Memory Corruption in - __ast_string_field_ptr_build_va (Reported by Arnd Schmitter) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-21178 - Improve documentation for manager command - Getvar, Setvar (Reported by Rusty Newton) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.12.0 * Mon Oct 20 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 11.11.0-1 - The Asterisk Development Team has announced the release of Asterisk 11.11.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.11.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-22551 - Session timer : UAS (Asterisk) starts counting - at Invite, UAC starts counting at 200 OK. (Reported by i2045) - * ASTERISK-23792 - Mutex left locked in chan_unistim.c (Reported - by Peter Whisker) - * ASTERISK-23582 - [patch]Inconsistent column length in *odbc - (Reported by Walter Doekes) - * ASTERISK-23803 - AMI action UpdateConfig EmptyCat clears all - categories but the requested one (Reported by zvision) - * ASTERISK-23035 - ConfBridge with name longer than max (32 chars) - results in several bridges with same conf_name (Reported by - Iñaki Cívico) - * ASTERISK-23824 - ConfBridge: Users cannot be muted via CLI or - AMI when waiting to enter a conference (Reported by Matt Jordan) - * ASTERISK-23683 - #includes - wildcard character in a path more - than one directory deep - results in no config parsing on module - reload (Reported by tootai) - * ASTERISK-23827 - autoservice thread doesn't exit at shutdown - (Reported by Corey Farrell) - * ASTERISK-23609 - Security: AMI action MixMonitor allows - arbitrary programs to be run (Reported by Corey Farrell) - * ASTERISK-23673 - Security: DOS by consuming the number of - allowed HTTP connections. (Reported by Richard Mudgett) - * ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite - a DEBUG level of zero (Reported by Rusty Newton) - * ASTERISK-23766 - [patch] Specify timeout for database write in - SQLite (Reported by Igor Goncharovsky) - * ASTERISK-23844 - Load of pbx_lua fails on sample extensions.lua - with Lua 5.2 or greater due to addition of goto statement - (Reported by Rusty Newton) - * ASTERISK-23818 - PBX_Lua: after asterisk startup module is - loaded, but dialplan not available (Reported by Dennis Guse) - * ASTERISK-23834 - res_rtp_asterisk debug message gives wrong - length if ICE (Reported by Richard Kenner) - * ASTERISK-23790 - [patch] - SIP From headers longer than 256 - characters result in dropped call and 'No closing bracket' - warnings. (Reported by uniken1) - * ASTERISK-23917 - res_http_websocket: Delay in client processing - large streams of data causes disconnect and stuck socket - (Reported by Matt Jordan) - * ASTERISK-23908 - [patch]When using FEC error correction, - asterisk tries considers negative sequence numbers as missing - (Reported by Torrey Searle) - * ASTERISK-23921 - refcounter.py uses excessive ram for large refs - files (Reported by Corey Farrell) - * ASTERISK-23948 - REF_DEBUG fails to record ao2_ref against - objects that were already freed (Reported by Corey Farrell) - * ASTERISK-23916 - [patch]SIP/SDP fmtp line may include whitespace - between attributes (Reported by Alexander Traud) - * ASTERISK-23984 - Infinite loop possible in ast_careful_fwrite() - (Reported by Steve Davies) - * ASTERISK-23897 - [patch]Change in SETUP ACK handling (checking - PI) in revision 413765 breaks working environments (Reported by - Pavel Troller) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-23492 - Add option to safe_asterisk to disable - backgrounding (Reported by Walter Doekes) - * ASTERISK-22961 - [patch] DTLS-SRTP not working with SHA-256 - (Reported by Jay Jideliov) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.11.0 * Thu Aug 28 2014 Jitka Plesnikova <jplesnik@redhat.com> - 11.10.2-2.2 - Perl 5.20 rebuild * Fri Aug 15 2014 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 11.10.2-2.1 - Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild * Thu Jun 19 2014 Jeffrey Ollie <jeff@ocjtech.us> - 11.10.2-2: - Drop the 389 directory server schema (1061414) * Thu Jun 19 2014 Jeffrey Ollie <jeff@ocjtech.us> - 11.10.2-1: - The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security - releases are released as versions 1.8.15-cert7, 11.6-cert4, 1.8.28.2, 11.10.2, - and 12.3.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - These releases resolve security vulnerabilities that were previously fixed in - 1.8.15-cert6, 11.6-cert3, 1.8.28.1, 11.10.1, and 12.3.1. Unfortunately, the fix - for AST-2014-007 inadvertently introduced a regression in Asterisk's TCP and TLS - handling that prevented Asterisk from sending data over these transports. This - regression and the security vulnerabilities have been fixed in the versions - specified in this release announcement. - - The security patches for AST-2014-007 have been updated with the fix for the - regression, and are available at http://downloads.asterisk.org/pub/security - - Please note that the release of these versions resolves the following security - vulnerabilities: - - * AST-2014-005: Remote Crash in PJSIP Channel Driver's Publish/Subscribe - Framework - - * AST-2014-006: Permission Escalation via Asterisk Manager User Unauthorized - Shell Access - - * AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP - Connections - - * AST-2014-008: Denial of Service in PJSIP Channel Driver Subscriptions - - For more information about the details of these vulnerabilities, please read - security advisories AST-2014-005, AST-2014-006, AST-2014-007, and AST-2014-008, - which were released with the previous versions that addressed these - vulnerabilities. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert7 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.28.2 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert4 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.10.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.3.2 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-005.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-006.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-007.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-008.pdf * Thu Jun 19 2014 Jeffrey Ollie <jeff@ocjtech.us> - 11.10.1-1: - The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security - releases are released as versions 1.8.15-cert6, 11.6-cert3, 1.8.28.1, 11.10.1, - and 12.3.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolves the following issue: - - * AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP - Connections - - Establishing a TCP or TLS connection to the configured HTTP or HTTPS port - respectively in http.conf and then not sending or completing a HTTP request - will tie up a HTTP session. By doing this repeatedly until the maximum number - of open HTTP sessions is reached, legitimate requests are blocked. - - Additionally, the release of 11.6-cert3, 11.10.1, and 12.3.1 resolves the - following issue: - - * AST-2014-006: Permission Escalation via Asterisk Manager User Unauthorized - Shell Access - - Manager users can execute arbitrary shell commands with the MixMonitor manager - action. Asterisk does not require system class authorization for a manager - user to use the MixMonitor action, so any manager user who is permitted to use - manager commands can potentially execute shell commands as the user executing - the Asterisk process. - - Additionally, the release of 12.3.1 resolves the following issues: - - * AST-2014-005: Remote Crash in PJSIP Channel Driver's Publish/Subscribe - Framework - - A remotely exploitable crash vulnerability exists in the PJSIP channel - driver's pub/sub framework. If an attempt is made to unsubscribe when not - currently subscribed and the endpoint's “sub_min_expiry” is set to zero, - Asterisk tries to create an expiration timer with zero seconds, which is not - allowed, so an assertion raised. - - * AST-2014-008: Denial of Service in PJSIP Channel Driver Subscriptions - - When a SIP transaction timeout caused a subscription to be terminated, the - action taken by Asterisk was guaranteed to deadlock the thread on which SIP - requests are serviced. Note that this behavior could only happen on - established subscriptions, meaning that this could only be exploited if an - attacker bypassed authentication and successfully subscribed to a real - resource on the Asterisk server. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2014-005, AST-2014-006, AST-2014-007, and AST-2014-008, - which were released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert6 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.28.1 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert3 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.10.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.3.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-005.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-006.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-007.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-008.pdf * Thu Jun 19 2014 Jeffrey Ollie <jeff@ocjtech.us> - 11.10.0-1: - The Asterisk Development Team has announced the release of Asterisk 11.10.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.10.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-23547 - [patch] app_queue removing callers from queue - when reloading (Reported by Italo Rossi) - * ASTERISK-23559 - app_voicemail fails to load after fix to - dialplan functions (Reported by Corey Farrell) - * ASTERISK-22846 - testsuite: masquerade super test fails on all - branches (still) (Reported by Matt Jordan) - * ASTERISK-23545 - Confbridge talker detection settings - configuration load bug (Reported by John Knott) - * ASTERISK-23546 - CB_ADD_LEN does not do what you'd think - (Reported by Walter Doekes) - * ASTERISK-23620 - Code path in app_stack fails to unlock list - (Reported by Bradley Watkins) - * ASTERISK-23616 - Big memory leak in logger.c (Reported by - ibercom) - * ASTERISK-23576 - Build failure on SmartOS / Illumos / SunOS - (Reported by Sebastian Wiedenroth) - * ASTERISK-23550 - Newer sound sets don't show up in menuselect - (Reported by Rusty Newton) - * ASTERISK-18331 - app_sms failure (Reported by David Woodhouse) - * ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by - Krzysztof Chmielewski) - * ASTERISK-23605 - res_http_websocket: Race condition in shutting - down websocket causes crash (Reported by Matt Jordan) - * ASTERISK-23707 - Realtime Contacts: Apparent mismatch between - PGSQL database state and Asterisk state (Reported by Mark - Michelson) - * ASTERISK-23381 - [patch]ChanSpy- Barge only works on the initial - 'spy', if the spied-on channel makes a new call, unable to - barge. (Reported by Robert Moss) - * ASTERISK-23665 - Wrong mime type for codec H263-1998 (h263+) - (Reported by Guillaume Maudoux) - * ASTERISK-23664 - Incorrect H264 specification in SDP. (Reported - by Guillaume Maudoux) - * ASTERISK-22977 - chan_sip+CEL: missing ANSWER and PICKUP event - for INVITE/w/replaces pickup (Reported by Walter Doekes) - * ASTERISK-23709 - Regression in Dahdi/Analog/waitfordialtone - (Reported by Steve Davies) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-23649 - [patch]Support for DTLS retransmission - (Reported by NITESH BANSAL) - * ASTERISK-23564 - [patch]TLS/SRTP status of channel not currently - available in a CLI command (Reported by Patrick Laimbock) - * ASTERISK-23754 - [patch] Use var/lib directory for log file - configured in asterisk.conf (Reported by Igor Goncharovsky) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.10.0 * Sat Jun 07 2014 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 11.9.0-2.1 - Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild * Thu May 15 2014 Dennis Gilmore <dennis@ausil.us> - 11.9.0-2 - build against gmime-devel not gmime22-devel - do not use -m64 on aarch64 * Wed Apr 23 2014 Jeffrey Ollie <jeff@ocjtech.us> - 11.9.0-1: - The Asterisk Development Team has announced the release of Asterisk 11.9.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.9.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-22790 - check_modem_rate() may return incorrect rate - for V.27 (Reported by Paolo Compagnini) - * ASTERISK-23034 - [patch] manager Originate doesn't abort on - failed format_cap allocation (Reported by Corey Farrell) - * ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in - sip.conf.sample (Reported by Eugene) - * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted - minus signs (Reported by Jeremy Lainé) - * ASTERISK-23046 - Custom CDR fields set during a GoSUB called - from app_queue are not inserted (Reported by Denis Pantsyrev) - * ASTERISK-23027 - [patch] Spelling typo "transfered" instead of - "transferred" (Reported by Jeremy Lainé) - * ASTERISK-23008 - Local channels loose CALLERID name when DAHDI - channel connects (Reported by Michael Cargile) - * ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted - request and request queue may differ - fix for locking (Reported - by adomjan) - * ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image - media offer due to invalid or unsupported syntax (Reported by - adomjan) - * ASTERISK-22861 - [patch]Specifying a null time as parameter to - GotoIfTime or ExecIfTime causes segmentation fault (Reported by - Sebastian Murray-Roberts) - * ASTERISK-17837 - extconfig.conf - Maximum Include level (1) - exceeded (Reported by pz) - * ASTERISK-22662 - Documentation fix? - queues.conf says - persistentmembers defaults to yes, it appears to lie (Reported - by Rusty Newton) - * ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot - handle selinux port restrictions (Reported by Corey Farrell) - * ASTERISK-23220 - STACK_PEEK function with no arguments causes - crash/core dump (Reported by James Sharp) - * ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload' - command multiple times on cli_aliases (Reported by Joel Vandal) - * ASTERISK-22757 - segfault in res_clialiases.so on reload when - mapping "module reload" command (Reported by Gareth Blades) - * ASTERISK-17727 - [patch] TLS doesn't get all certificate chain - (Reported by LN) - * ASTERISK-23178 - devicestate.h: device state setting functions - are documented with the wrong return values (Reported by - Jonathan Rose) - * ASTERISK-23232 - LocalBridge AMI Event LocalOptimization value - is opposite to what's expected (Reported by Leon Roy) - * ASTERISK-23098 - [patch]possible null pointer dereference in - format.c (Reported by Marcello Ceschia) - * ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if - res_parking.so is not loaded, or if res_parking.conf has no - configuration (Reported by CJ Oster) - * ASTERISK-23069 - Custom CDR variable not recorded when set in - macro called from app_queue (Reported by Bryan Anderson) - * ASTERISK-19499 - ConfBridge MOH is not working for transferee - after attended transfer (Reported by Timo Teräs) - * ASTERISK-23261 - [patch]Output mixup in - ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686) - * ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic - payload change in rtp mapping in the 200 OK response (Reported - by NITESH BANSAL) - * ASTERISK-23255 - UUID included for Redhat, but missing for - Debian distros in install_prereq script (Reported by Rusty - Newton) - * ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR - variables for subsequent records (Reported by zvision) - * ASTERISK-23141 - Asterisk crashes on Dial(), in - pbx_find_extension at pbx.c (Reported by Maxim) - * ASTERISK-23336 - Asterisk warning "Don't know how to indicate - condition 33 on ooh323c" on outgoing calls from H323 to SIP peer - (Reported by Alexander Semych) - * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set - to minrate=2400, then res_fax refuse to load (Reported by David - Brillert) - * ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set - - probably introduced in 11.7.0 (Reported by OK) - * ASTERISK-23323 - [patch]chan_sip: missing p->owner checks in - handle_response_invite (Reported by Walter Doekes) - * ASTERISK-23406 - [patch]Fix typo in "sip show peer" (Reported by - ibercom) - * ASTERISK-23310 - bridged channel crashes in bridge_p2p_rtp_write - (Reported by Jeremy Lainé) - * ASTERISK-22911 - [patch]Asterisk fails to resume WebRTC call - from hold (Reported by Vytis Valentinavičius) - * ASTERISK-23104 - Specifying the SetVar AMI without a Channel - cause Asterisk to crash (Reported by Joel Vandal) - * ASTERISK-21930 - [patch]WebRTC over WSS is not working. - (Reported by John) - * ASTERISK-23383 - Wrong sense test on stat return code causes - unchanged config check to break with include files. (Reported by - David Woolley) - * ASTERISK-20149 - Crash when faxing SIP to SIP with strictrtp set - to yes (Reported by Alexandr Gordeev) - * ASTERISK-17523 - Qualify for static realtime peers does not work - (Reported by Maciej Krajewski) - * ASTERISK-21406 - [patch] chan_sip deadlock on monlock between - unload_module and do_monitor (Reported by Corey Farrell) - * ASTERISK-23373 - [patch]Security: Open FD exhaustion with - chan_sip Session-Timers (Reported by Corey Farrell) - * ASTERISK-23340 - Security Vulnerability: stack allocation of - cookie headers in loop allows for unauthenticated remote denial - of service attack (Reported by Matt Jordan) - * ASTERISK-23311 - Manager - MoH Stop Event fails to show up when - leaving Conference (Reported by Benjamin Keith Ford) - * ASTERISK-23420 - [patch]Memory leak in manager_add_filter - function in manager.c (Reported by Etienne Lessard) - * ASTERISK-23488 - Logic error in callerid checksum processing - (Reported by Russ Meyerriecks) - * ASTERISK-23461 - Only first user is muted when joining - confbridge with 'startmuted=yes' (Reported by Chico Manobela) - * ASTERISK-20841 - fromdomain not honored on outbound INVITE - request (Reported by Kelly Goedert) - * ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f) - at astobj2.c:120 (Reported by Jamuel Starkey) - * ASTERISK-23509 - [patch]SayNumber for Polish language tries to - play empty files for numbers divisible by 100 (Reported by - zvision) - * ASTERISK-23103 - [patch]Crash in ast_format_cmp, in ao2_find - (Reported by JoshE) - * ASTERISK-23391 - Audit dialplan function usage of channel - variable (Reported by Corey Farrell) - * ASTERISK-23548 - POST to ARI sometimes returns no body on - success (Reported by Scott Griepentrog) - * ASTERISK-23460 - ooh323 channel stuck if call is placed directly - and gatekeeper is not available (Reported by Dmitry Melekhov) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-22980 - [patch]Allow building cdr_radius and cel_radius - against libfreeradius-client (Reported by Jeremy Lainé) - * ASTERISK-22661 - Unable to exit ChanSpy if spied channel does - not have a call in progress (Reported by Chris Hillman) - * ASTERISK-23099 - [patch] WSS: enable ast_websocket_read() - function to read the whole available data at first and then wait - for any fragmented packets (Reported by Thava Iyer) * Tue Mar 11 2014 Jeffrey Ollie <jeff@ocjtech.us> - 11.8.1-1: - The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security - releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1, - and 12.1.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following issues: - - * AST-2014-001: Stack overflow in HTTP processing of Cookie headers. - - Sending a HTTP request that is handled by Asterisk with a large number of - Cookie headers could overflow the stack. - - Another vulnerability along similar lines is any HTTP request with a - ridiculous number of headers in the request could exhaust system memory. - - * AST-2014-002: chan_sip: Exit early on bad session timers request - - This change allows chan_sip to avoid creation of the channel and - consumption of associated file descriptors altogether if the inbound - request is going to be rejected anyway. - - Additionally, the release of 12.1.1 resolves the following issue: - - * AST-2014-003: res_pjsip: When handling 401/407 responses don't assume a - request will have an endpoint. - - This change removes the assumption that an outgoing request will always - have an endpoint and makes the authenticate_qualify option work once again. - - Finally, a security advisory, AST-2014-004, was released for a vulnerability - fixed in Asterisk 12.1.0. Users of Asterisk 12.0.0 are encouraged to upgrade to - 12.1.1 to resolve both vulnerabilities. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2014-001, AST-2014-002, AST-2014-003, and AST-2014-004, - which were released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert5 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.26.1 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.8.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.1.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2014-001.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-002.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-003.pdf - * http://downloads.asterisk.org/pub/security/AST-2014-004.pdf * Tue Mar 4 2014 Jeffrey Ollie <jeff@ocjtech.us> - 11.8.0-1: - The Asterisk Development Team has announced the release of Asterisk 11.8.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.8.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - Bugs fixed in this release: - ----------------------------------- - * ASTERISK-22544 - Italian prompt vm-options has advertisement in - it (Reported by Rusty Newton) - * ASTERISK-21383 - STUN Binding Requests Not Being Sent Back from - Asterisk to Chrome (Reported by Shaun Clark) - * ASTERISK-22478 - [patch]Can't use pound(hash) symbol for custom - DTMF menus in ConfBridge (processed as directive) (Reported by - Nicolas Tanski) - * ASTERISK-12117 - chan_sip creates a new local tag (from-tag) for - every register message (Reported by Pawel Pierscionek) - * ASTERISK-20862 - Asterisk min and max member penalties not - honored when set with 0 (Reported by Schmooze Com) - * ASTERISK-22746 - [patch]Crash in chan_dahdi during caller id - read (Reported by Michael Walton) - * ASTERISK-22788 - [patch] main/translate.c: access to variable f - after free in ast_translate() (Reported by Corey Farrell) - * ASTERISK-21242 - Segfault when T.38 re-invite retransmission - receives 200 OK (Reported by Ashley Winters) - * ASTERISK-22590 - BufferOverflow in unpacksms16() when receiving - 16 bit multipart SMS with app_sms (Reported by Jan Juergens) - * ASTERISK-22905 - Prevent Asterisk functions that are 'dangerous' - from being executed from external interfaces (Reported by Matt - Jordan) - * ASTERISK-23021 - Typos in code : "avaliable" instead of - "available" (Reported by Jeremy Lainé) - * ASTERISK-22970 - [patch]Documentation fix for QUOTE() (Reported - by Gareth Palmer) - * ASTERISK-21960 - ooh323 channels stuck (Reported by Dmitry - Melekhov) - * ASTERISK-22350 - DUNDI - core dump on shutdown - segfault in - sqlite3_reset from /usr/lib/libsqlite3.so.0 (Reported by Birger - "WIMPy" Harzenetter) - * ASTERISK-22942 - [patch] - Asterisk crashed after - Set(FAXOPT(faxdetect)=t38) (Reported by adomjan) - * ASTERISK-22856 - [patch]SayUnixTime in polish reads minutes - instead of seconds (Reported by Robert Mordec) - * ASTERISK-22854 - [patch] - Deadlock between cel_pgsql unload and - core_event_dispatcher taskprocessor thread (Reported by Etienne - Lessard) - * ASTERISK-22910 - [patch] - REPLACE() calls strcpy on overlapping - memory when <replace-char> is empty (Reported by Gareth Palmer) - * ASTERISK-22871 - cel_pgsql module not loading after "reload" or - "reload cel_pgsql.so" command (Reported by Matteo) - * ASTERISK-23084 - [patch]rasterisk needlessly prints the - AST-2013-007 warning (Reported by Tzafrir Cohen) - * ASTERISK-17138 - [patch] Asterisk not re-registering after it - receives "Forbidden - wrong password on authentication" - (Reported by Rudi) - * ASTERISK-23011 - [patch]configure.ac and pbx_lua don't support - lua 5.2 (Reported by George Joseph) - * ASTERISK-22834 - Parking by blind transfer when lot full orphans - channels (Reported by rsw686) - * ASTERISK-23047 - Orphaned (stuck) channel occurs during a failed - SIP transfer to parking space (Reported by Tommy Thompson) - * ASTERISK-22946 - Local From tag regression with sipgate.de - (Reported by Stephan Eisvogel) - * ASTERISK-23010 - No BYE message sent when sip INVITE is received - (Reported by Ryan Tilton) - * ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set - - probably introduced in 11.7.0 (Reported by OK) - - Improvements made in this release: - ----------------------------------- - * ASTERISK-22728 - [patch] Improve Understanding Of 'Forcerport' - When Running "sip show peers" (Reported by Michael L. Young) - * ASTERISK-22659 - Make a new core and extra sounds release - (Reported by Rusty Newton) - * ASTERISK-22919 - core show channeltypes slicing (Reported by - outtolunc) - * ASTERISK-22918 - dahdi show channels slices PRI channel dnid on - output (Reported by outtolunc) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.8.0 * Sat Dec 28 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.7.0-1: - The Asterisk Development Team has announced the release of Asterisk 11.7.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.7.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following is a sample of the issues resolved in this release: - - * --- app_confbridge: Can now set the language used for announcements - to the conference. - (Closes issue ASTERISK-19983. Reported by Jonathan White) - - * --- app_queue: Fix CLI "queue remove member" queue_log entry. - (Closes issue ASTERISK-21826. Reported by Oscar Esteve) - - * --- chan_sip: Do not increment the SDP version between 183 and 200 - responses. - (Closes issue ASTERISK-21204. Reported by NITESH BANSAL) - - * --- chan_sip: Allow a sip peer to accept both AVP and AVPF calls - (Closes issue ASTERISK-22005. Reported by Torrey Searle) - - * --- chan_sip: Fix Realtime Peer Update Problem When Un-registering - And Expires Header In 200ok - (Closes issue ASTERISK-22428. Reported by Ben Smithurst) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.7.0 * Sat Dec 28 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.6.1-1: - The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security - releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, - 10.12.4-digiumphones, and 11.6.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following issues: - - * A buffer overflow when receiving odd length 16 bit messages in app_sms. An - infinite loop could occur which would overwrite memory when a message is - received into the unpacksms16() function and the length of the message is an - odd number of bytes. - - * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk - now marks certain individual dialplan functions as 'dangerous', which will - inhibit their execution from external sources. - - A 'dangerous' function is one which results in a privilege escalation. For - example, if one were to read the channel variable SHELL(rm -rf /) Bad - Things(TM) could happen; even if the external source has only read - permissions. - - Execution from external sources may be enabled by setting 'live_dangerously' - to 'yes' in the [options] section of asterisk.conf. Although doing so is not - recommended. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2013-006 and AST-2013-007, which were - released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert4 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.2-cert3 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.24.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.4 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.4-digiumphones - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf - * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf * Sat Dec 28 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.6.0-1: - The Asterisk Development Team has announced the release of Asterisk 11.6.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.6.0 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: - - * --- Confbridge: empty conference not being torn down - (Closes issue ASTERISK-21859. Reported by Chris Gentle) - - * --- Let Queue wrap up time influence member availability - (Closes issue ASTERISK-22189. Reported by Tony Lewis) - - * --- Fix a longstanding issue with MFC-R2 configuration that - prevented users - (Closes issue ASTERISK-21117. Reported by Rafael Angulo) - - * --- chan_iax2: Fix saving the wrong expiry time in astdb. - (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) - - * --- Fix segfault for certain invalid WebSocket input. - (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 * Mon Oct 21 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.5.1-3: - Disable hardened build, as it's apparently causing problems loading modules. * Thu Aug 29 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.5.1-2: - Enable hardened build BZ#954338 - Significant clean ups * Thu Aug 29 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.5.1-1: - The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases - are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones, - and 11.5.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following issues: - - * A remotely exploitable crash vulnerability exists in the SIP channel driver if - an ACK with SDP is received after the channel has been terminated. The - handling code incorrectly assumes that the channel will always be present. - - * A remotely exploitable crash vulnerability exists in the SIP channel driver if - an invalid SDP is sent in a SIP request that defines media descriptions before - connection information. The handling code incorrectly attempts to reference - the socket address information even though that information has not yet been - set. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2013-004 and AST-2013-005, which were - released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert3 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.2-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.23.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.3 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.3-digiumphones - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.5.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2013-004.pdf - * http://downloads.asterisk.org/pub/security/AST-2013-005.pdf - - The Asterisk Development Team has announced the release of Asterisk 11.5.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.5.0 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 Segfault In app_queue When "persistentmembers" Is Enabled - And Using Realtime - (Closes issue ASTERISK-21738. Reported by JoshE) - - * --- IAX2: fix race condition with nativebridge transfers. - (Closes issue ASTERISK-21409. Reported by alecdavis) - - * --- Fix The Payload Being Set On CN Packets And Do Not Set Marker - Bit - (Closes issue ASTERISK-21246. Reported by Peter Katzmann) - - * --- Fix One-Way Audio With auto_* NAT Settings When SIP Calls - Initiated By PBX - (Closes issue ASTERISK-21374. Reported by Michael L. Young) - - * --- chan_sip: NOTIFYs for BLF start queuing up and fail to be sent - out after retries fail - (Closes issue ASTERISK-21677. Reported by Dan Martens) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.5.0 * Sat Aug 03 2013 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 11.4.0-2.2 - Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild * Wed Jul 17 2013 Petr Pisar <ppisar@redhat.com> - 11.4.0-2.1 - Perl 5.18 rebuild * Fri May 24 2013 Rex Dieter <rdieter@fedoraproject.org> 11.4.0-2 - rebuild (libical) * Mon May 20 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.4.0-1: - The Asterisk Development Team has announced the release of Asterisk 11.4.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.4.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following is a sample of the issues resolved in this release: - - * --- Fix Sorting Order For Parking Lots Stored In Static Realtime - (Closes issue ASTERISK-21035. Reported by Alex Epshteyn) - - * --- Fix StopMixMonitor Hanging Up When Unable To Stop MixMonitor On - A Channel - (Closes issue ASTERISK-21294. Reported by daroz) - - * --- When a session timer expires during a T.38 call, re-invite with - correct SDP - (Closes issue ASTERISK-21232. Reported by Nitesh Bansal) - - * --- Fix white noise on SRTP decryption - (Closes issue ASTERISK-21323. Reported by andrea) - - * --- Fix reload skinny with active devices. - (Closes issue ASTERISK-16610. Reported by wedhorn) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.4.0 * Fri May 10 2013 Tom Callaway <spot@fedoraproject.org> - 11.3.0-2: - fix build with lua 5.2 * Tue Apr 23 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.3.0-1: - The Asterisk Development Team has announced the release of Asterisk 11.3.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.3.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following is a sample of the issues resolved in this release: - - * --- Fix issue where chan_mobile fails to bind to first available - port - (Closes issue ASTERISK-16357. Reported by challado) - - * --- Fix Queue Log Reporting Every Call COMPLETECALLER With "h" - Extension Present - (Closes issue ASTERISK-20743. Reported by call) - - * --- Retain XMPP filters across reconnections so external modules - continue to function as expected. - (Closes issue ASTERISK-20916. Reported by kuj) - - * --- Ensure that a declined media stream is terminated with a '\r\n' - (Closes issue ASTERISK-20908. Reported by Dennis DeDonatis) - - * --- Fix pjproject compilation in certain circumstances - (Closes issue ASTERISK-20681. Reported by Dinesh Ramjuttun) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.3.0 * Thu Mar 28 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.2.2-1: - The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases - are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones, - and 11.2.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following issues: - - * A possible buffer overflow during H.264 format negotiation. The format - attribute resource for H.264 video performs an unsafe read against a media - attribute when parsing the SDP. - - This vulnerability only affected Asterisk 11. - - * A denial of service exists in Asterisk's HTTP server. AST-2012-014, fixed - in January of this year, contained a fix for Asterisk's HTTP server for a - remotely-triggered crash. While the fix prevented the crash from being - triggered, a denial of service vector still exists with that solution if an - attacker sends one or more HTTP POST requests with very large Content-Length - values. - - This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11 - - * A potential username disclosure exists in the SIP channel driver. When - authenticating a SIP request with alwaysauthreject enabled, allowguest - disabled, and autocreatepeer disabled, Asterisk discloses whether a user - exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways. - - This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11 - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2013-001, AST-2013-002, and AST-2013-003, which were - released at the same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.20.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.2-digiumphones - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.2.2 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2013-001.pdf - * http://downloads.asterisk.org/pub/security/AST-2013-002.pdf - * http://downloads.asterisk.org/pub/security/AST-2013-003.pdf * Sun Feb 10 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.2.1-1: - The Asterisk Development Team has announced the release of Asterisk 11.2.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.2.1 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - * --- Fix astcanary startup problem due to wrong pid value from before - daemon call - (Closes issue ASTERISK-20947. Reported by Jakob Hirsch) - - * --- Update init.d scripts to handle stderr; readd splash screen for - remote consoles - (Closes issue ASTERISK-20945. Reported by Warren Selby) - - * --- Reset RTP timestamp; sequence number on SSRC change - (Closes issue ASTERISK-20906. Reported by Eelco Brolman) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.1 * Fri Jan 18 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.2.0-1: - The Asterisk Development Team has announced the release of Asterisk 11.2.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.2.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following is a sample of the issues resolved in this release: - - * --- app_meetme: Fix channels lingering when hung up under certain - conditions - (Closes issue ASTERISK-20486. Reported by Michael Cargile) - - * --- Fix stuck DTMF when bridge is broken. - (Closes issue ASTERISK-20492. Reported by Jeremiah Gowdy) - - * --- Add missing support for "who hung up" to chan_motif. - (Closes issue ASTERISK-20671. Reported by Matt Jordan) - - * --- Remove a fixed size limitation for producing SDP and change how - ICE support is disabled by default. - (Closes issue ASTERISK-20643. Reported by coopvr) - - * --- Fix chan_sip websocket payload handling - (Closes issue ASTERISK-20745. Reported by Iñaki Baz Castillo) - - * --- Fix pjproject compilation in certain circumstances - (Closes issue ASTERISK-20681. Reported by Dinesh Ramjuttun) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.0 * Thu Jan 3 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.1.2-1: - The Asterisk Development Team has announced a security release for Asterisk 11, - Asterisk 11.1.2. This release addresses the security vulnerabilities reported in - AST-2012-014 and AST-2012-015, and replaces the previous version of Asterisk 11 - released for these security vulnerabilities. The prior release left open a - vulnerability in res_xmpp that exists only in Asterisk 11; as such, other - versions of Asterisk were resolved correctly by the previous releases. - - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following two issues: - - * Stack overflows that occur in some portions of Asterisk that manage a TCP - connection. In SIP, this is exploitable via a remote unauthenticated session; - in XMPP and HTTP connections, this is exploitable via remote authenticated - sessions. The vulnerabilities in SIP and HTTP were corrected in a prior - release of Asterisk; the vulnerability in XMPP is resolved in this release. - - * A denial of service vulnerability through exploitation of the device state - cache. Anonymous calls had the capability to create devices in Asterisk that - would never be disposed of. Handling the cachability of device states - aggregated via XMPP is handled in this release. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-014 and AST-2012-015. - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.2 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-014.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-015.pdf - - Thank you for your continued support of Asterisk - and we apologize for having - to do this twice! * Wed Jan 2 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.1.1-1: - The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.11 and Asterisk 1.8, 10, and 11. The available security releases - are released as versions 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones, - and 11.1.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of these versions resolve the following two issues: - - * Stack overflows that occur in some portions of Asterisk that manage a TCP - connection. In SIP, this is exploitable via a remote unauthenticated session; - in XMPP and HTTP connections, this is exploitable via remote authenticated - sessions. - - * A denial of service vulnerability through exploitation of the device state - cache. Anonymous calls had the capability to create devices in Asterisk that - would never be disposed of. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-014 and AST-2012-015, which were released at the - same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert10 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.19.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.11.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.11.1-digiumphones - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-014.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-015.pdf * Wed Dec 12 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.1.0-1: - The Asterisk Development Team has announced the release of Asterisk 11.1.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.1.0 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 execution of 'i' extension due to uninitialized variable. - (Closes issue ASTERISK-20455. Reported by Richard Miller) - - * --- Prevent resetting of NATted realtime peer address on reload. - (Closes issue ASTERISK-18203. Reported by daren ferreira) - - * --- Fix ConfBridge crash if no timing module loaded. - (Closes issue ASTERISK-19448. Reported by feyfre) - - * --- Fix the Park 'r' option when a channel parks itself. - (Closes issue ASTERISK-19382. Reported by James Stocks) - - * --- Fix an issue where outgoing calls would fail to establish audio - due to ICE negotiation failures. - (Closes issue ASTERISK-20554. Reported by mmichelson) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.1.0 * Fri Dec 7 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.2-1: - The Asterisk Development Team has announced the release of Asterisk 11.0.2. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.0.2 resolves an issue reported by the - community and would have not been possible without your participation. - Thank you! - - The following is the issue resolved in this release: - - * --- chan_local: Fix local_pvt ref leak in local_devicestate(). - (Closes issue ASTERISK-20769. Reported by rmudgett) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.2 * Wed Dec 5 2012 Dan Horák <dan[at]danny.cz> - 11.0.1-3 - simplify LDFLAGS setting * Fri Nov 30 2012 Dennis Gilmore <dennis@ausil.us> - 11.0.1-2 - clean up things to allow building on arm arches * Mon Nov 5 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.1-1 - The Asterisk Development Team has announced the release of Asterisk 11.0.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.0.1 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - * --- chan_sip: Fix a bug causing SIP reloads to remove all entries - from the registry - (Closes issue ASTERISK-20611. Reported by Alisher) - - * --- confbridge: Fix a bug which made conferences not record with - AMI/CLI commands - (Closes issue ASTERISK-20601. Reported by Vilius) - - * --- Fix an issue with res_http_websocket where the chan_sip - WebSocket handler could not be registered. - (Closes issue ASTERISK-20631. Reported by danjenkins) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.1 * Tue Oct 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.0-1: - The Asterisk Development Team is pleased to announce the release of - Asterisk 11.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - Asterisk 11 is the next major release series of Asterisk. It is a Long Term - Support (LTS) release, similar to Asterisk 1.8. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - For important information regarding upgrading to Asterisk 11, please see the - Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 - - A short list of new features includes: - - * A new channel driver named chan_motif has been added which provides support - for Google Talk and Jingle in a single channel driver. This new channel - driver includes support for both audio and video, RFC2833 DTMF, all codecs - supported by Asterisk, hold, unhold, and ringing notification. It is also - compliant with the current Jingle specification, current Google Jingle - specification, and the original Google Talk protocol. - - * Support for the WebSocket transport for chan_sip. - - * SIP peers can now be configured to support negotiation of ICE candidates. - - * The app_page application now no longer depends on DAHDI or app_meetme. It - has been re-architected to use app_confbridge internally. - - * Hangup handlers can be attached to channels using the CHANNEL() function. - Hangup handlers will run when the channel is hung up similar to the h - extension; however, unlike an h extension, a hangup handler is associated with - the actual channel and will execute anytime that channel is hung up, - regardless of where it is in the dialplan. - - * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial - allows you to execute a dialplan subroutine on a channel before a call is - placed but after the application performing a dial action is invoked. This - means that the handlers are executed after the creation of the callee - channels, but before any actions have been taken to actually dial the callee - channels. - - * Log messages can now be easily associated with a certain call by looking at - a new unique identifier, "Call Id". Call ids are attached to log messages for - just about any case where it can be determined that the message is related - to a particular call. - - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in - Asterisk. Unlike traditional ACLs defined in specific module configuration - files, Named ACLs can be shared across multiple modules. - - * The Hangup Cause family of functions and dialplan applications allow for - inspection of the hangup cause codes for each channel involved in a call. - This allows a dialplan writer to determine, for each channel, who hung up and - for what reason(s). - - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() - lets you set some of the configuration options from the general section - of features.conf on a per-channel basis. FEATUREMAP() lets you customize - the key sequence used to activate built-in features, such as blindxfer, - and automon. - - * Support for DTLS-SRTP in chan_sip. - - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups - and callgroups to be defined for several channel drivers. - - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework. - - More information about the new features can be found on the Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation - - A full list of all new features can also be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES - - For a full list of changes in the current release, please see the ChangeLog. - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0 * Wed Oct 17 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.0-0.7.rc2: - The Asterisk Development Team has announced the second release candidate of - Asterisk 11.0.0. This release candidate is available for immediate - download at http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 11.0.0-rc2 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 candidate: - - * --- Fix an issue where outgoing calls would fail to establish audio - due to ICE negotiation failures. - (Closes issue ASTERISK-20554. Reported by mmichelson) - - * --- Ensure Asterisk fails TCP/TLS SIP calls when certificate - checking fails - (Closes issue ASTERISK-20559. Reported by kmoore) - - * --- Don't make chan_sip export global symbols. - (Closes issue ASTERISK-20545. Reported by kmoore) - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.0-rc2 * Tue Oct 9 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.0-0.6.rc1 - The Asterisk Development Team is pleased to announce the first release candidate - of Asterisk 11.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - All interested users of Asterisk are encouraged to participate in the - Asterisk 11 testing process. Please report any issues found to the issue - tracker, https://issues.asterisk.org/jira. It is also very useful to see - successful test reports. Please post those to the asterisk-dev mailing list. - All Asterisk users are invited to participate in the #asterisk-testing channel - on IRC to work together in testing the many parts of Asterisk. - - Asterisk 11 is the next major release series of Asterisk. It will be a Long - Term Support (LTS) release, similar to Asterisk 1.8. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - For important information regarding upgrading to Asterisk 11, please see the - Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 - - A short list of new features includes: - - * A new channel driver named chan_motif has been added which provides support - for Google Talk and Jingle in a single channel driver. This new channel - driver includes support for both audio and video, RFC2833 DTMF, all codecs - supported by Asterisk, hold, unhold, and ringing notification. It is also - compliant with the current Jingle specification, current Google Jingle - specification, and the original Google Talk protocol. - - * Support for the WebSocket transport for chan_sip. - - * SIP peers can now be configured to support negotiation of ICE candidates. - - * The app_page application now no longer depends on DAHDI or app_meetme. It - has been re-architected to use app_confbridge internally. - - * Hangup handlers can be attached to channels using the CHANNEL() function. - Hangup handlers will run when the channel is hung up similar to the h - extension; however, unlike an h extension, a hangup handler is associated with - the actual channel and will execute anytime that channel is hung up, - regardless of where it is in the dialplan. - - * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial - allows you to execute a dialplan subroutine on a channel before a call is - placed but after the application performing a dial action is invoked. This - means that the handlers are executed after the creation of the callee - channels, but before any actions have been taken to actually dial the callee - channels. - - * Log messages can now be easily associated with a certain call by looking at - a new unique identifier, "Call Id". Call ids are attached to log messages for - just about any case where it can be determined that the message is related - to a particular call. - - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in - Asterisk. Unlike traditional ACLs defined in specific module configuration - files, Named ACLs can be shared across multiple modules. - - * The Hangup Cause family of functions and dialplan applications allow for - inspection of the hangup cause codes for each channel involved in a call. - This allows a dialplan writer to determine, for each channel, who hung up and - for what reason(s). - - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() - lets you set some of the configuration options from the general section - of features.conf on a per-channel basis. FEATUREMAP() lets you customize - the key sequence used to activate built-in features, such as blindxfer, - and automon. - - * Support for DTLS-SRTP in chan_sip. - - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups - and callgroups to be defined for several channel drivers. - - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework. - - More information about the new features can be found on the Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation - - A full list of all new features can also be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES - - For a full list of changes in the current release, please see the ChangeLog. - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-rc1 * Wed Sep 26 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.0-0.5.beta2 - Don't forget format_ilbc module * Wed Sep 26 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.0-0.4.beta2 - The Asterisk Development Team is pleased to announce the second beta release of - Asterisk 11.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - All interested users of Asterisk are encouraged to participate in the - Asterisk 11 testing process. Please report any issues found to the issue - tracker, https://issues.asterisk.org/jira. It is also very useful to see - successful test reports. Please post those to the asterisk-dev mailing list. - All Asterisk users are invited to participate in the #asterisk-testing channel - on IRC to work together in testing the many parts of Asterisk. - - Asterisk 11 is the next major release series of Asterisk. It will be a Long - Term Support (LTS) release, similar to Asterisk 1.8. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - For important information regarding upgrading to Asterisk 11, please see the - Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 - - A short list of new features includes: - - * A new channel driver named chan_motif has been added which provides support - for Google Talk and Jingle in a single channel driver. This new channel - driver includes support for both audio and video, RFC2833 DTMF, all codecs - supported by Asterisk, hold, unhold, and ringing notification. It is also - compliant with the current Jingle specification, current Google Jingle - specification, and the original Google Talk protocol. - - * Support for the WebSocket transport for chan_sip. - - * SIP peers can now be configured to support negotiation of ICE candidates. - - * The app_page application now no longer depends on DAHDI or app_meetme. It - has been re-architected to use app_confbridge internally. - - * Hangup handlers can be attached to channels using the CHANNEL() function. - Hangup handlers will run when the channel is hung up similar to the h - extension; however, unlike an h extension, a hangup handler is associated with - the actual channel and will execute anytime that channel is hung up, - regardless of where it is in the dialplan. - - * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial - allows you to execute a dialplan subroutine on a channel before a call is - placed but after the application performing a dial action is invoked. This - means that the handlers are executed after the creation of the callee - channels, but before any actions have been taken to actually dial the callee - channels. - - * Log messages can now be easily associated with a certain call by looking at - a new unique identifier, "Call Id". Call ids are attached to log messages for - just about any case where it can be determined that the message is related - to a particular call. - - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in - Asterisk. Unlike traditional ACLs defined in specific module configuration - files, Named ACLs can be shared across multiple modules. - - * The Hangup Cause family of functions and dialplan applications allow for - inspection of the hangup cause codes for each channel involved in a call. - This allows a dialplan writer to determine, for each channel, who hung up and - for what reason(s). - - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() - lets you set some of the configuration options from the general section - of features.conf on a per-channel basis. FEATUREMAP() lets you customize - the key sequence used to activate built-in features, such as blindxfer, - and automon. - - * Support for DTLS-SRTP in chan_sip. - - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups - and callgroups to be defined for several channel drivers. - - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework. - - More information about the new features can be found on the Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation - - A full list of all new features can also be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES - - For a full list of changes in the current release, please see the ChangeLog. - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta2 * Wed Sep 26 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.8.0-1 - The Asterisk Development Team has announced the release of Asterisk 10.8.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.8.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following is a sample of the issues resolved in this release: - - * --- AST-2012-012: Resolve AMI User Unauthorized Shell Access through - ExternalIVR - (Closes issue ASTERISK-20132. Reported by Zubair Ashraf of IBM X-Force Research) - - * --- AST-2012-013: Resolve ACL rules being ignored during calls by - some IAX2 peers - (Closes issue ASTERISK-20186. Reported by Alan Frisch) - - * --- Handle extremely out of order RFC 2833 DTMF - (Closes issue ASTERISK-18404. Reported by Stephane Chazelas) - - * --- Resolve severe memory leak in CEL logging modules. - (Closes issue AST-916. Reported by Thomas Arimont) - - * --- Only re-create an SRTP session when needed - (Issue ASTERISK-20194. Reported by Nicolo Mazzon) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.8.0 * Tue Sep 04 2012 Dan Horák <dan[at]danny.cz> - 11.0.0-0.3.beta1 - fix build on s390 * Tue Sep 04 2012 Dan Horák <dan[at]danny.cz> - 10.7.1-2 - fix build on s390 * Thu Aug 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.7.1-1 - The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are - released as versions 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones - resolve the following two issues: - - * A permission escalation vulnerability in Asterisk Manager Interface. This - would potentially allow remote authenticated users the ability to execute - commands on the system shell with the privileges of the user running the - Asterisk application. Please note that the README-SERIOUSLY.bestpractices.txt - file delivered with Asterisk has been updated due to this and other related - vulnerabilities fixed in previous versions of Asterisk. - - * When an IAX2 call is made using the credentials of a peer defined in a - dynamic Asterisk Realtime Architecture (ARA) backend, the ACL rules for that - peer are not applied to the call attempt. This allows for a remote attacker - who is aware of a peer's credentials to bypass the ACL rules set for that - peer. - - These issues and their resolutions are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-012 and AST-2012-013, which were released at the - same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert7 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.15.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.7.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.7.1-digiumphones - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-012.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-013.pdf * Thu Aug 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.7.0-1 - The Asterisk Development Team has announced the release of Asterisk 10.7.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.7.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following is a sample of the issues resolved in this release: - - * --- Fix deadlock potential with ast_set_hangupsource() calls. - (Closes issue ASTERISK-19801. Reported by Alec Davis) - - * --- Fix request routing issue when outboundproxy is used. - (Closes issue ASTERISK-20008. Reported by Marcus Hunger) - - * --- Set the Caller ID "tag" on peers even if remote party - information is present. - (Closes issue ASTERISK-19859. Reported by Thomas Arimont) - - * --- Fix NULL pointer segfault in ast_sockaddr_parse() - (Closes issue ASTERISK-20006. Reported by Michael L. Young) - - * --- Do not perform install on existing directories - (Closes issue ASTERISK-19492. Reported by Karl Fife) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.7.0 * Thu Aug 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.6.1-1 - The Asterisk Development Team has announced the release of Asterisk 10.6.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.6.1 resolves an issue reported by the - community and would have not been possible without your participation. - Thank you! - - The following is the issue resolved in this release: - - * --- Remove a superfluous and dangerous freeing of an SSL_CTX. - (Closes issue ASTERISK-20074. Reported by Trevor Helmsley) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.6.1 * Thu Aug 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.6.0-1 - The Asterisk Development Team has announced the release of Asterisk 10.6.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.6.0 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: - - * --- format_mp3: Fix a possible crash in mp3_read(). - (Closes issue ASTERISK-19761. Reported by Chris Maciejewsk) - - * --- Fix local channel chains optimizing themselves out of a call. - (Closes issue ASTERISK-16711. Reported by Alec Davis) - - * --- Re-add LastMsgsSent value for SIP peers - (Closes issue ASTERISK-17866. Reported by Steve Davies) - - * --- Prevent sip_pvt refleak when an ast_channel outlasts its - corresponding sip_pvt. - (Closes issue ASTERISK-19425. Reported by David Cunningham) - - * --- Send more accurate identification information in dialog-info SIP - NOTIFYs. - (Closes issue ASTERISK-16735. Reported by Maciej Krajewski) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.6.0 * Sat Aug 18 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.0-0.2.beta1 - The Asterisk Development Team is pleased to announce the first beta release of - Asterisk 11.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - All interested users of Asterisk are encouraged to participate in the - Asterisk 11 testing process. Please report any issues found to the issue - tracker, https://issues.asterisk.org/jira. It is also very useful to see - successful test reports. Please post those to the asterisk-dev mailing list. - All Asterisk users are invited to participate in the #asterisk-testing channel - on IRC to work together in testing the many parts of Asterisk. - - Asterisk 11 is the next major release series of Asterisk. It will be a Long - Term Support (LTS) release, similar to Asterisk 1.8. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - For important information regarding upgrading to Asterisk 11, please see the - Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 - - A short list of new features includes: - - * A new channel driver named chan_motif has been added which provides support - for Google Talk and Jingle in a single channel driver. This new channel - driver includes support for both audio and video, RFC2833 DTMF, all codecs - supported by Asterisk, hold, unhold, and ringing notification. It is also - compliant with the current Jingle specification, current Google Jingle - specification, and the original Google Talk protocol. - - * Support for the WebSocket transport for chan_sip. - - * SIP peers can now be configured to support negotiation of ICE candidates. - - * The app_page application now no longer depends on DAHDI or app_meetme. It - has been re-architected to use app_confbridge internally. - - * Hangup handlers can be attached to channels using the CHANNEL() function. - Hangup handlers will run when the channel is hung up similar to the h - extension; however, unlike an h extension, a hangup handler is associated with - the actual channel and will execute anytime that channel is hung up, - regardless of where it is in the dialplan. - - * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial - allows you to execute a dialplan subroutine on a channel before a call is - placed but after the application performing a dial action is invoked. This - means that the handlers are executed after the creation of the caller/callee - channels, but before any actions have been taken to actually dial the callee - channels. - - * Log messages can now be easily associated with a certain call by looking at - a new unique identifier, "Call Id". Call ids are attached to log messages for - just about any case where it can be determined that the message is related - to a particular call. - - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in - Asterisk. Unlike traditional ACLs defined in specific module configuration - files, Named ACLs can be shared across multiple modules. - - * The Hangup Cause family of functions and dialplan applications allow for - inspection of the hangup cause codes for each channel involved in a call. - This allows a dialplan writer to determine, for each channel, who hung up and - for what reason(s). - - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() - lets you set some of the configuration options from the general section - of features.conf on a per-channel basis. FEATUREMAP() lets you customize - the key sequence used to activate built-in features, such as blindxfer, - and automon. - - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups - and callgroups to be defined for several channel drivers. - - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework. - - More information about the new features can be found on the Asterisk wiki: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation - - A full list of all new features can also be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES - - For a full list of changes in the current release, please see the ChangeLog. - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta1 * Wed Jul 18 2012 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 10.5.2-1.2 - Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild * Mon Jul 09 2012 Petr Pisar <ppisar@redhat.com> - 10.5.2-1.1 - Perl 5.16 rebuild * Thu Jul 5 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.5.2-1: - The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are - released as versions 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones - resolve the following two issues: - - * If Asterisk sends a re-invite and an endpoint responds to the re-invite with - a provisional response but never sends a final response, then the SIP dialog - structure is never freed and the RTP ports for the call are never released. If - an attacker has the ability to place a call, they could create a denial of - service by using all available RTP ports. - - * If a single voicemail account is manipulated by two parties simultaneously, - a condition can occur where memory is freed twice causing a crash. - - These issues and their resolution are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-010 and AST-2012-011, which were released at the - same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert4 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.13.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.2-digiumphones - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-010.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-011.pdf * Thu Jun 28 2012 Petr Pisar <ppisar@redhat.com> - 10.5.1-1.1 - Perl 5.16 rebuild * Fri Jun 15 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.5.1-1 - The Asterisk Development Team has announced a security release for Asterisk 10. - This security release is released as version 10.5.1. - - The release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 10.5.1 resolves the following issue: - - * A remotely exploitable crash vulnerability was found in the Skinny (SCCP) - Channel driver. When an SCCP client sends an Off Hook message, followed by - a Key Pad Button Message, a structure that was previously set to NULL is - dereferenced. This allows remote authenticated connections the ability to - cause a crash in the server, denying services to legitimate users. - - This issue and its resolution is described in the security advisory. - - For more information about the details of this vulnerability, please read - security advisory AST-2012-009, 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-10.5.1 - - The security advisory is available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-009.pdf * Fri Jun 15 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.5.0-1 - The Asterisk Development Team has announced the release of Asterisk 10.5.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.5.0 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: - - * --- Turn off warning message when bind address is set to any. - (Closes issue ASTERISK-19456. Reported by Michael L. Young) - - * --- Prevent overflow in calculation in ast_tvdiff_ms on 32-bit - machines - (Closes issue ASTERISK-19727. Reported by Ben Klang) - - * --- Make DAHDISendCallreroutingFacility wait 5 seconds for a reply - before disconnecting the call. - (Closes issue ASTERISK-19708. Reported by mehdi Shirazi) - - * --- Fix recalled party B feature flags for a failed DTMF atxfer. - (Closes issue ASTERISK-19383. Reported by lgfsantos) - - * --- Fix DTMF atxfer running h exten after the wrong bridge ends. - (Closes issue ASTERISK-19717. Reported by Mario) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.5.0 * Mon Jun 11 2012 Petr Pisar <ppisar@redhat.com> - 10.4.2-1.1 - Perl 5.16 rebuild * Wed May 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.4.2-1 - The Asterisk Development Team has announced the release of Asterisk 10.4.2. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.4.2 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: - - * --- Resolve crash in subscribing for MWI notifications - (Closes issue ASTERISK-19827. Reported by B. R) - - * --- Fix crash in ConfBridge when user announcement is played for - more than 2 users - (Closes issue ASTERISK-19899. Reported by Florian Gilcher) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.4.2 * Wed May 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.4.1-1 - The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are - released as versions 1.8.11-cert2, 1.8.12.1, and 10.4.1. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases - - The release of Asterisk 1.8.11-cert2, 1.8.12.1, and 10.4.1 resolve the following - two issues: - - * A remotely exploitable crash vulnerability exists in the IAX2 channel - driver if an established call is placed on hold without a suggested music - class. Asterisk will attempt to use an invalid pointer to the music - on hold class name, potentially causing a crash. - - * A remotely exploitable crash vulnerability was found in the Skinny (SCCP) - Channel driver. When an SCCP client closes its connection to the server, - a pointer in a structure is set to NULL. If the client was not in the - on-hook state at the time the connection was closed, this pointer is later - dereferenced. This allows remote authenticated connections the ability to - cause a crash in the server, denying services to legitimate users. - - These issues and their resolution are described in the security advisories. - - For more information about the details of these vulnerabilities, please read - security advisories AST-2012-007 and AST-2012-008, which were released at the - same time as this announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.11-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.12.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.4.1 - - The security advisories are available at: - - * http://downloads.asterisk.org/pub/security/AST-2012-007.pdf - * http://downloads.asterisk.org/pub/security/AST-2012-008.pdf * Fri May 4 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.4.0-1 - The Asterisk Development Team has announced the release of Asterisk 10.4.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk - - The release of Asterisk 10.4.0 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-10.4.0 * Tue Apr 24 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.3.1-1 - 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 * Thu Mar 29 2012 Russell Bryant <russell@russellbryant.net> - 10.3.0-1 - Update to 10.3.0 * Fri Mar 16 2012 Russell Bryant <russell@russellbryant.net> - 10.2.1-1 - Update to 10.2.1 from upstream. - Fix remote stack overflow in app_milliwatt. - Fix remote stack overflow, including possible code injection, in HTTP digest authentication handling. - Disable asterisk-corosync package, as it doesn't build right now. - Resolves: rhbz#804045, rhbz#804038, rhbz#804042 * Thu Feb 16 2012 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.1.2-2 - * Add patch extracted from upstream to build with Corosync since - OpenAIS is no longer available. - * Add PrivateTmp=true to systemd service file (#782478) - * Add some macros to make it easier to build with fewer dependencies - (with corresponding less functionality) (#787389) - * Add isa macros in a few places plus a few other changes to make it - easier to cross-compile. (#787779) * Thu Feb 16 2012 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.1.2-1 - The Asterisk Development Team has announced the release of Asterisk 10.1.2. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 10.1.2 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following are the issues resolved in this release: - - * --- Fix SIP INFO DTMF handling for non-numeric codes --- - (Closes issue ASTERISK-19290. Reported by: Ira Emus) - - * --- Fix crash in ParkAndAnnounce --- - (Closes issue ASTERISK-19311. Reported-by: tootai) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.2 * Thu Feb 16 2012 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.1.1-1 - The Asterisk Development Team has announced the release of Asterisk 10.1.1. This - release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 10.1.1 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following is a sample of the issues resolved in this release: - - * --- Fixes deadlocks occuring in chan_agent --- - (Closes issue ASTERISK-19285. Reported by: Alex Villacis Lasso) - - * --- Ensure entering T.38 passthrough does not cause an infinite loop --- - (Closes issue ASTERISK-18951. Reported-by: Kristijan Vrban) - - For a full list of changes in this release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.1 * Thu Feb 16 2012 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.1.0-1 - The Asterisk Development Team is pleased to announce the release of - Asterisk 10.1.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 10.1.0 resolves several issues reported by the - community and would have not been possible without your participation. - Thank you! - - The following is a sample of the issues resolved in this release: - - * AST-2012-001: prevent crash when an SDP offer - is received with an encrypted video stream when support for video - is disabled and res_srtp is loaded. (closes issue ASTERISK-19202) - Reported by: Catalin Sanda - - * Allow playback of formats that don't support seeking. ast_streamfile - previously did unconditional seeking on files that broke playback of - formats that don't support that functionality. This patch avoids the - seek that was causing the problem. - (closes issue ASTERISK-18994) Patched by: Timo Teras - - * Add pjmedia probation concepts to res_rtp_asterisk's learning mode. In - order to better handle RTP sources with strictrtp enabled (which is the - default setting in 10) using the learning mode to figure out new sources - when they change is handled by checking for a number of consecutive (by - sequence number) packets received to an rtp struct based on a new - configurable value called 'probation'. Also, during learning mode instead - of liberally accepting all packets received, we now reject packets until a - clear source has been determined. - - * Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop. Failing - to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop - causes the loop to exit prematurely. This causes a variety of negative side - effects, depending on when the loop exits. This patch handles the frame by - essentially swallowing the frame in the local loop, as the current channel - drivers expect the RTP bridge to handle the frame, and, in the case of the - local bridge loop, no additional action is necessary. - (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested - by: Matt Jordan - - * Fix timing source dependency issues with MOH. Prior to this patch, - res_musiconhold existed at the same module priority level as the timing - sources that it depends on. This would cause a problem when music on - hold was reloaded, as the timing source could be changed after - res_musiconhold was processed. This patch adds a new module priority - level, AST_MODPRI_TIMING, that the various timing modules are now loaded - at. This now occurs before loading other resource modules, such - that the timing source is guaranteed to be set prior to resolving - the timing source dependencies. - (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, - Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont - Patched by elguero - - * Fix RTP reference leak. If a blind transfer were initiated using a - REFER without a prior reINVITE to place the call on hold, AND if Asterisk - were sending RTCP reports, then there was a reference leak for the - RTP instance of the transferrer. - (closes issue ASTERISK-19192) Reported by: Tyuta Vitali - - * Fix blind transfers from failing if an 'h' extension - is present. This prevents the 'h' extension from being run on the - transferee channel when it is transferred via a native transfer - mechanism such as SIP REFER. (closes issue ASTERISK-19173) Reported - by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by - Mark Michelson (license 5049) - - * Restore call progress code for analog ports. Extracting sig_analog - from chan_dahdi lost call progress detection functionality. Fix - analog ports from considering a call answered immediately after - dialing has completed if the callprogress option is enabled. - (closes issue ASTERISK-18841) - Reported by: Richard Miller Patched by Richard Miller - - * Fix regression that 'rtp/rtcp set debup ip' only works when a port - was also specified. - (closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by: - Walter Doekes - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.0 * Thu Feb 16 2012 Russell Bryant <russellb@fedoraproject.org> - 10.0.0-2 - Remove asterisk-ais. OpenAIS was removed from Fedora. * Thu Jan 12 2012 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 10.0.0-1.1 - Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild * Tue Jan 3 2012 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-1 - Don't build API docs as the build never finishes * Thu Dec 15 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-1 - The Asterisk Development Team is proud to announce the release of - Asterisk 10.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - Asterisk 10 is the next major release series of Asterisk. It will be a - Standard support release, similar to Asterisk 1.6.2. For more information about - support time lines for Asterisk releases, see the Asterisk versions page: - - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - With the release of the Asterisk 10 branch, the preceding '1.' has been removed - from the version number per the blog post available at - - - http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ - - The release of Asterisk 10 would not have been possible without the support and - contributions of the community. - - You can find an overview of the work involved with the 10.0.0 release in the - summary: - - http://svn.asterisk.org/svn/asterisk/tags/10.0.0/asterisk-10.0.0-summary.txt - - A short list of available features includes: - - * T.38 gateway functionality has been added to res_fax. - * Protocol independent out-of-call messaging support. Text messages not - associated with an active call can now be routed through the Asterisk - dialplan. SIP and XMPP are supported so far. - * New highly optimized and customizable ConfBridge application capable of mixing - audio at sample rates ranging from 8kHz-192kHz - * Addition of video_mode option in confbridge.conf to provide basic video - conferencing in the ConfBridge() dialplan application. - * Support for defining hints has been added to pbx_lua. - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svn.asterisk.org/svn/asterisk/branches/10/CHANGES - - Also, when upgrading a system between major versions, it is imperative that you - read and understand the contents of the UPGRADE.txt file, which is located at: - - http://svn.asterisk.org/svn/asterisk/branches/10/UPGRADE.txt * Fri Dec 9 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-0.7.rc3 - The Asterisk Development Team has announced the third release candidate of - Asterisk 10.0.0. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 10.0.0-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: - - * Add ASTSBINDIR to the list of configurable paths - - This patch also makes astdb2sqlite3 and astcanary use the configured - directory instead of relying on $PATH. - - * Don't crash on INFO automon request with no channel - - AST-2011-014. When automon was enabled in features.conf, it was possible - to crash Asterisk by sending an INFO request if no channel had been - created yet. - - * Fixed crash from orphaned MWI subscriptions in chan_sip - - This patch resolves the issue where MWI subscriptions are orphaned - by subsequent SIP SUBSCRIBE messages. - - * Fix a change in behavior in 'database show' from 1.8. - - In 1.8 and previous versions, one could use any fullword portion of - the key name, including the full key, to obtain the record. Until this - patch, this did not work for the full key. - - * Default to nat=yes; warn when nat in general and peer differ - - AST-2011-013. It is possible to enumerate SIP usernames when the general and - user/peer nat settings differ in whether to respond to the port a request is - sent from or the port listed for responses in the Via header. In 1.4 and - 1.6.2, this would mean if one setting was nat=yes or nat=route and the other - was either nat=no or nat=never. In 1.8 and 10, this would mean when one - was nat=force_rport and the other was nat=no. - - In order to address this problem, it was decided to switch the default - behavior to nat=yes/force_rport as it is the most commonly used option - and to strongly discourage setting nat per-peer/user when at all - possible. - - * Fixed SendMessage stripping extension from To: header in SIP MESSAGE - - When using the MessageSend application to send a SIP MESSAGE to a - non-peer, chan_sip stripped off the extension and failed to add it back - to the sip_pvt structure before transmitting. This patch adds the full - URI passed in from the message core to the sip_pvt structure. - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.0.0-rc3 * Wed Nov 16 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-0.6.rc2 - The Asterisk Development Team has announced the second release candidate of - Asterisk 10.0.0. This release candidate is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - The release of Asterisk 10.0.0-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: - - * Ensure that a null vmexten does not cause a segfault - - * Fix issue with ConfBridge participants hanging up during DTMF feature - menu usage getting stuck in conference forever - (closes issue ASTERISK-18829) - Reported by: zvision - - * Fix app_macro.c MODULEINFO section termination - (closes issue ASTERISK-18848) - Reported by: Tony Mountifield - - For a full list of changes in this release candidate, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.0.0-rc2 * Fri Nov 11 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-0.5.rc1 - The Asterisk Development Team is pleased to announce the first release candidate - of Asterisk 10.0.0. This release candidate is available for immediate download - at http://downloads.asterisk.org/pub/telephony/asterisk/ - - All Asterisk users are encouraged to participate in the Asterisk 10 testing - process. Please report any issues found to the issue tracker, - https://issues.asterisk.org/jira. It is also very useful to see successful test - reports. Please post those to the asterisk-dev mailing list. - - All Asterisk users are invited to participate in the #asterisk-testing - channel on IRC to work together in testing the many parts of Asterisk. - - Asterisk 10 is the next major release series of Asterisk. It will be a - Standard support release, similar to Asterisk 1.6.2. For more - information about support time lines for Asterisk releases, see the Asterisk - versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - A short list of features includes: - - * T.38 gateway functionality has been added to res_fax. - * Protocol independent out-of-call messaging support. Text messages not - associated with an active call can now be routed through the Asterisk - dialplan. SIP and XMPP are supported so far. - * New highly optimized and customizable ConfBridge application capable of mixing - audio at sample rates ranging from 8kHz-192kHz - (More information available at - https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 ) - * Addition of video_mode option in confbridge.conf to provide basic video - conferencing in the ConfBridge() dialplan application. - * Support for defining hints has been added to pbx_lua. - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/10/CHANGES - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-rc1 * Tue Oct 18 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-0.4.beta2 - Add patch from upstream SVN to fix AST-2011-012 * Fri Oct 14 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-0.3.beta2 - Patch cleanup day * Thu Sep 29 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-0.2.beta2 - The Asterisk Development Team is pleased to announce the second beta release of - Asterisk 10.0.0. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - With the release of the Asterisk 10 branch, the preceding '1.' has been removed - from the version number per the blog post available at - http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ - - All interested users of Asterisk are encouraged to participate in the - Asterisk 10 testing process. Please report any issues found to the issue - tracker, https://issues.asterisk.org/jira. It is also very useful to see - successful test reports. Please post those to the asterisk-dev mailing list. - - All Asterisk users are invited to participate in the #asterisk-testing - channel on IRC to work together in testing the many parts of Asterisk. - - Asterisk 10 is the next major release series of Asterisk. It will be a - Standard support release, similar to Asterisk 1.6.2. For more - information about support time lines for Asterisk releases, see the Asterisk - versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - A short list of features includes: - - * T.38 gateway functionality has been added to res_fax. - - * Protocol independent out-of-call messaging support. Text messages not - associated with an active call can now be routed through the Asterisk - dialplan. SIP and XMPP are supported so far. - - * New highly optimized and customizable ConfBridge application capable of mixing - audio at sample rates ranging from 8kHz-192kHz - - * Addition of video_mode option in confbridge.conf to provide basic video - conferencing in the ConfBridge() dialplan application. - - * Support for defining hints has been added to pbx_lua. - - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). - - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svnview.digium.com/svn/asterisk/branches/10/CHANGES - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-beta2 * Mon Jul 25 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-0.1.beta1 - - The Asterisk Development Team is pleased to announce the first beta release of - Asterisk 10.0.0-beta1. This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - With the release of the Asterisk 10 branch, the preceding '1.' has been removed - from the version number per the blog post available at - http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ - - All interested users of Asterisk are encouraged to participate in the - Asterisk 10 testing process. Please report any issues found to the issue - tracker, https://issues.asterisk.org/jira. It is also very useful to see - successful test reports. Please post those to the asterisk-dev mailing list. - - All Asterisk users are invited to participate in the #asterisk-testing - channel on IRC to work together in testing the many parts of Asterisk. - Additionally users can make use of the RPM and DEB packages now being built for - all Asterisk releases. More information available at - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages - - Asterisk 10 is the next major release series of Asterisk. It will be a - Standard support release, similar to Asterisk 1.6.2. For more - information about support time lines for Asterisk releases, see the Asterisk - versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions - - A short list of included features includes: - - * T.38 gateway functionality has been added to res_fax. - * Protocol independent out-of-call messaging support. Text messages not - associated with an active call can now be routed through the Asterisk - dialplan. SIP and XMPP are supported so far. - * New highly optimized and customizable ConfBridge application capable of mixing - audio at sample rates ranging from 8kHz-192kHz - * Addition of video_mode option in confbridge.conf to provide basic video - conferencing in the ConfBridge() dialplan application. - * Support for defining hints has been added to pbx_lua. - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB). - * Much, much more! - - A full list of new features can be found in the CHANGES file. - - http://svn.digium.com/view/asterisk/branches/10/CHANGES?view=checkout - - For a full list of changes in the current release, please see the ChangeLog: - - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-beta1 * Thu Jul 21 2011 Petr Sabata <contyk@redhat.com> - 1.8.5.0-1.2 - Perl mass rebuild * Wed Jul 20 2011 Petr Sabata <contyk@redhat.com> - 1.8.5.0-1.1 - Perl mass rebuild * Mon Jul 11 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.5-0.2 - Rebuild for net-snmp 5.7 * Fri Jul 1 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.5-0.1.rc1 - Fix systemd dependencies in EL6 and F15 * Thu Jun 30 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.4.4-2 - Fedora Directory Server -> 389 Directory Server * Wed Jun 29 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.4.3-3 - Don't forget stereorize * Mon Jun 27 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.4.2-2 - Convert to systemd * Fri Jun 17 2011 Marcela Mašláňová <mmaslano@redhat.com> - 1.8.4.2-1.2 - Perl mass rebuild * Fri Jun 10 2011 Marcela Mašláňová <mmaslano@redhat.com> - 1.8.4.2-1.1 - Perl 5.14 mass rebuild * Fri Jun 3 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.3.2-2 - Bump release and rebuild for mysql 5.5.10 soname change. * Thu Mar 17 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.3-0.6.rc2 - Bump release to build for F15 * Wed Feb 9 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3-0.5.rc2 - Remove isa macros * Wed Feb 9 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3-0.4.rc2 - Make library dependencies architecture specific * Mon Feb 07 2011 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 1.8.3-0.3.rc2 - Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild * Wed Jan 26 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.2.2-2 - Build with SRTP support * Mon Jan 24 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <dennis@ausil.us> - 1.8.0-6 - dont package up the ices bits on el the client doesnt exist for us * Tue Jan 18 2011 Dennis Gilmore <dennis@ausil.us> - 1.8.0-5 - dont build the 389 directory server package its not available on rhel6 * Fri Dec 10 2010 Dennis Gilmore <dennis@ausil.us> - 1.8.0-4 - dont always build AIS modules we dont have the BuildRequires on epel * Fri Oct 29 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-3 - Rebuild for new net-snmp. * Tue Oct 26 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-2 - Always build AIS modules * Thu Oct 21 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.8.0-0.2.beta2 - Rebuild against libpri 1.4.12 * Mon Aug 2 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.2.8-0.3.rc1 - Add patch to remove requirement on latex2html * Tue Jun 01 2010 Marcela Maslanova <mmaslano@redhat.com> - 1.6.2.8-0.2.rc1 - Mass rebuild with perl-5.12.0 * Tue May 4 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.2.7-0.2.rc3 - Update to 1.6.2.7-rc3 * Thu Apr 15 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.7-0.1.rc2 - Update to 1.6.2.7-rc2 * Fri Mar 12 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.2.6-0.1.rc2 - Update to 1.6.2.6-rc2 * Mon Mar 8 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.5-2 - Add a patch that fixes CLI history when linking against external libedit. * Thu Feb 25 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.2.1-0.1.rc1 - Update to 1.6.2.1-rc1 * Sat Dec 19 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-1 - Released version of 1.6.2.0 * Wed Dec 9 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.16.rc8 - Update to 1.6.2.0-rc8 * Wed Dec 2 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.15.rc7 - Update to 1.6.2.0-rc7 * Tue Dec 1 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.2.0-0.12.rc6 - Update to 1.6.2.0-rc6 * Mon Nov 9 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.11.rc5 - Update to 1.6.2.0-rc5 * Fri Nov 6 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.10.rc4 - Update to 1.6.2.0-rc4 * Tue Oct 27 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.2.0-0.6.rc3 - Require latex2html used in static-http documents * Wed Oct 7 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.5.rc3 - Change ownership and permissions on config files to protect them. * Tue Oct 6 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.4.rc3 - Update to 1.6.2.0-rc3 * Wed Sep 30 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.3.rc2 - Merge firmware subpackage back into the main package. * Wed Sep 30 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.1-0.26.rc1 - Add a patch from Quentin Armitage and rebuld. * Fri Aug 21 2009 Tomas Mraz <tmraz@redhat.com> - 1.6.1-0.25.rc1 - rebuilt with new openssl * Fri Jul 24 2009 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 1.6.1-0.24.rc1 - Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild * Thu Mar 5 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <rel-eng@lists.fedoraproject.org> - 1.6.1-0.22.rc1 - Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild * Sun Feb 8 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <tmraz@redhat.com> - 1.6.1-0.13.beta4 - rebuild with new openssl * Sun Jan 4 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.12.beta4 - Fedora Directory Server compatibility patch/subpackage. * Sun Jan 4 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.10.beta4 - Fix up paths. BZ#477238 * Sat Jan 3 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.9.beta4 - Update patches * Sat Jan 3 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.8.beta4 - Update to 1.6.1-beta4 * Tue Dec 9 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.7.beta3 - Update to 1.6.1-beta3 * Tue Dec 9 2008 Alex Lancaster <alexlan[AT]fedoraproject org> - 1.6.1-0.6.beta2 - Rebuild for new gmime * Fri Nov 7 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.5.beta2 - Add patch to fix missing variable on PPC. * Fri Nov 7 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.4.beta2 - Update PPC systems don't have sys/io.h patch. * Fri Nov 7 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.3.beta2 - PPC systems don't have sys/io.h * Fri Nov 7 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.2.beta2 - Update to 1.6.1 beta 2 * Wed Nov 5 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0.1-3 - Fix issue with init script giving wrong path to config file. * Thu Oct 16 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.0-1 - Update to final release. * Thu Sep 11 2008 - Bastien Nocera <bnocera@redhat.com> - 1.6.0-0.22.beta9 - Rebuild * Wed Jul 30 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.0-0.20.beta9 - Bump release and rebuild with new libpri and zaptel. * Fri Jul 25 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.0-0.18.beta9 - Add patch for LDAP extracted from upstream SVN (#442011) * Wed Jul 2 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.0-0.15.beta9 - Bump release and rebuild to fix libtds breakage. * Mon May 19 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.6.0-0.9.beta6 - Add Postgresql schemas from contrib as documentation to the Postgresql subpackage. * Tue Mar 25 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <tcallawa@redhat.com> - 1.6.0-0.5.beta5 - add Requires for versioned perl (libperl.so) * Wed Mar 5 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.4.beta5 - Update to 1.6.0-beta5 - Remove upstreamed patches. * Mon Mar 3 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.3.beta4 - Package the directory used to store monitor recordings. * Tue Feb 26 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.2.beta4 - Add patch from David Woodhouse that fixes building on PPC64. * Tue Feb 26 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.1.beta4 - Update to 1.6.0 beta 4 * Wed Feb 13 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.17-1 - Update to 1.4.17 to fix AST-2008-001. * Fri Dec 28 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.16.2-1 - Update to 1.4.16.2 * Sat Dec 22 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.16.1-2 - Bump release and rebuild to fix broken dep on uw-imap. * Wed Dec 19 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.16.1-1 - Update to the real 1.4.16.1. * Wed Dec 19 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.16-1 - Update to 1.4.16 to fix security bug. * Sat Dec 15 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-7 - Really, really fix the build problems on devel. * Sat Dec 15 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-6 - Tweaks to get to build on x86_64 * Wed Dec 12 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-5 - Exclude PPC64 * Wed Dec 12 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.15-3 - Really get rid of zero length map files. * Mon Dec 10 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-2 - Get rid of zero length map files. - Shorten descriptions of voicemail subpackages * Fri Nov 30 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-1 - Update to 1.4.15 * Tue Nov 20 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.14-2 - Fix license and other rpmlint warnings. * Mon Nov 19 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.14-1 - Update to 1.4.14 * Fri Nov 16 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.13-7 - Add chan_mobile * Tue Nov 13 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.13-1 - Update to 1.4.13 * Tue Oct 9 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.12.1-1 - Update to 1.4.12.1 * Wed Aug 22 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.11-1 - Update to 1.4.11 * Fri Aug 10 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.10.1-1 - Update to 1.4.10.1. * Tue Aug 7 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.10-1 - Update to 1.4.10 (security update). * Tue Aug 7 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.9-7 - Add a patch that allows alternate extensions to be defined in users.conf * Mon Aug 6 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.9-6 - Update app_conference patch. Enter/leave sounds are now possible. * Fri Jul 27 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.9-4 - Don't build app_mp3 * Wed Jul 25 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.9-3 - Add app_conference * Wed Jul 25 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.9-1 - Update to 1.4.9 * Tue Jul 17 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.8-1 - Update to 1.4.8 - Drop ixjuser patch. * Tue Jul 10 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.7.1-1 - Update to 1.4.7.1 * Mon Jul 9 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.7-1 - Update to 1.4.7 - RxFAX/TxFAX applications * Sun Jul 1 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.6-4 - It's "sbin", not "bin" silly. * Sat Jun 30 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.6-3 - Add patch that lets us change TOS bits even when running non-root * Fri Jun 29 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.6-2 - voicemail needs to require /usr/bin/sox and /usr/bin/sendmail * Fri Jun 29 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.6-1 - Update to 1.4.6 - Remove upstreamed patch. * Thu Jun 21 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.5-9 - Move sounds * Mon Jun 18 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-8 - Update some more ownership/permissions * Mon Jun 18 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-7 - Fix some permissions. * Mon Jun 18 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-6 - Update init script patch - Move pid file to subdir of /var/run * Mon Jun 18 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-5 - Update init script patch to run as non-root * Sun Jun 17 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-4 - Build modules that depend on FreeTDS. - Don't build voicemail with ODBC storage. * Sun Jun 17 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-3 - Have the build output the commands executing, rather than covering them up. * Fri Jun 15 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-1 - Update to 1.4.5 - Remove upstreamed patch. * Wed May 9 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.4-2 - Add a patch to fix CVE-2007-2488/ASA-2007-013 * Fri Apr 27 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.4-1 - Update to 1.4.4 * Wed Mar 21 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.2-1 - Update to 1.4.2 * Tue Mar 6 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.1-2 - Package the IAXy firmware - Minor clean-ups in files * Mon Mar 5 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.0-6.beta4 - Update to 1.4.0-beta4 - Various cleanups. * Fri Oct 20 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.0-4.beta3 - Remove explicit dependency on glibc-kernheaders. - Build jabber modules on PPC * Wed Oct 18 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.4.0-2.beta3 - Remove requirement on asterisk-sounds-core until licensing can be figured out. * Wed Oct 18 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.0-1.beta3 - Update to 1.4.0-beta3 * Sun Oct 15 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.0-0.beta2 - Update to 1.4.0-beta2 * Tue Jul 25 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.10-1 - Update to 1.2.10. * Wed Jun 7 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.9.1 - Update to 1.2.9.1 * Fri Jun 2 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.2.7.1-5 - Build mISDN channel drivers, modelled after spec file from David Woodhouse * Thu Apr 20 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.2.7.1-3 - Add chan_bluetooth * Wed Apr 19 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.7.1-2 - Split off more stuff into subpackages. * Wed Apr 12 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.7-1 - Update to 1.2.7 * Mon Apr 10 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.2.6-1 - Update to 1.2.6 * Mon Mar 6 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.2.4-4 - Actually apply the patch. * Mon Feb 13 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.2.4-2 - BR sqlite2-devel * Tue Jan 31 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.4-1 - Update to 1.2.4. * Wed Jan 25 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 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 <jeff@ocjtech.us> - 1.2.3-1 - First version for Fedora Extras.
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