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File asterisk.spec of Package asterisk (Revision 37)
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# # spec file for package asterisk (Version 1.4.21) # # # rootforbuild %define build_h323 0 Name: asterisk BuildRequires: alsa-devel curl-devel expat gcc-c++ libgsm-devel libgsm1 libpri libtiff-devel postgresql-devel spandsp speex-devel unixODBC-devel wget BuildRequires: libpri >= 1.4.1 BuildRequires: zaptel >= 1.4.1 BuildRequires: mISDN mISDNuser BuildRequires: alsa-lib-devel BuildRequires: bison BuildRequires: bluez-libs BuildRequires: doxygen BuildRequires: libogg-devel BuildRequires: termcap BuildRequires: libvorbis-devel BuildRequires: m4 BuildRequires: ncurses-devel BuildRequires: net-snmp-devel BuildRequires: newt-devel BuildRequires: openssl-devel BuildRequires: spandsp spandsp-devel BuildRequires: autoconf openldap2 openldap2-devel Buildrequires: iksemel unixODBC termcap iksemel-devel Buildrequires: postgresql-devel postgresql-server %if %build_h323 == 1 BuildRequires: cyrus-sasl-devel BuildRequires: openh323-devel BuildRequires: pwlib-devel BuildRequires: openldap2-devel BuildRequires: SDL-devel %endif Requires: newt ncurses postgresql-libs speex openldap2 postgresql-devel libgsm1 Requires: iksemel %ifnarch s390 s390x BuildRequires: zaptel >= 1.4.1 %endif URL: http://www.asterisk.org Summary: The Asterisk Open Source PBX Version: 1.4.21 Release: 1 License: BSD License and BSD-like, GNU General Public License (GPL) PreReq: /usr/sbin/useradd Group: Productivity/Telephony/Servers Source0: %name-%version.tar.gz Source1: %name-spandsp.tar.gz Source2: openh323-v1_19_0_1-src-tar.bz2 BuildRoot: %{_tmppath}/%{name}-%{version}-build Patch0: asterisk-1.4-h323.patch #Patch1: asterisk-1.4-chan_h323-moduleinfo.patch Patch2: asterisk-1.4-guiconfchanges.patch Patch4: asterisk.patch %description Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides voicemail services with directory, call conferencing, interactive voice response, and call queuing. It has support for three-way calling, caller ID services, ADSI, SIP, and H.323 (as both client and gateway). Documentation is available on the Asterisk home page (http://www.asterisk.org) and on the Asterisk wiki (http://www.voip-info.org/wiki-Asterisk). Authors: -------- Mark Spencer <markster@digium.com> #%debug_package %package alsa Summary: Soundcard Module for Asterisk Group: Productivity/Telephony/Servers PreReq: asterisk %description alsa This package allows Asterisk to use a soundcard supported by ALSA as a telephone. Authors: -------- Mark Spencer <markster@digium.com> %if %build_h323 == 1 %package h323 Summary: Voice over IP Module for Asterisk Group: Productivity/Telephony/Servers PreReq: asterisk %description h323 This package adds support for the H.323 voice over IP (VoIP) protocol to Asterisk. Support for the SIP and IAX2 protocols is included in the asterisk base package. Authors: -------- Mark Spencer <markster@digium.com> %endif %package odbc Summary: Database Module for Asterisk Group: Productivity/Telephony/Servers PreReq: asterisk %description odbc This package allows Asterisk to use read configuration data from, and write call logs to ODBC databases. Authors: -------- Mark Spencer <markster@digium.com> %if ! 0%{?suse_version} == 1010 %package pgsql Summary: Database Module for Asterisk Group: Productivity/Telephony/Servers PreReq: asterisk %description pgsql This package allows Asterisk to use read configuration data from, and write call logs to PostgreSQL databases. Authors: -------- Mark Spencer <markster@digium.com> %endif %package spandsp Summary: Softfax Module for Asterisk Group: Productivity/Telephony/Servers PreReq: asterisk %description spandsp This package allows Asterisk to send/receive faxes. Authors: -------- Steve Underwood <steveu@coppice.org> %package zaptel Summary: Telephony Hardware Module for Asterisk Group: Productivity/Telephony/Servers PreReq: asterisk %description zaptel This module allows Asterisk to use telephony hardware that is supported by the zaptel kernel drivers. Supported hardware ranges from FXO and FXS cards over ISDN BRI cards to T1, and E1 cards with up to four interfaces. See the zaptel package for details. Authors: -------- Mark Spencer <markster@digium.com> %package devel Summary: Development files for Asterisk Group: Development/Libraries PreReq: asterisk %description devel Development files for Asterisk Authors: -------- Mark Spencer <markster@digium.com> %prep %setup -n asterisk-%{version} -a 1 -b 2 echo -e "\n" | contrib/scripts/get_ilbc_source.sh %if %build_h323 == 1 %patch0 -p1 -b .h323rpm #%patch1 -p1 -b .h323moduleinfo %patch2 -p1 -b .guiconf %endif %patch4 -p0 %build ./bootstrap.sh %if %build_h323 == 1 export configh323="--with-h323=%{_includedir}/openh323" %endif %configure $configh323 %{__make} menuselect.makeopts %{__make} %{?_smp_mflags} %install rm -rf %{buildroot} %{__make} install DESTDIR=%{buildroot} %{__make} samples DESTDIR=%{buildroot} %{__install} -p -m 0644 include/asterisk.h %{buildroot}%{_includedir} %{__install} -D -p -m 0755 contrib/init.d/rc.suse.asterisk %{buildroot}%{_initrddir}/asterisk # get rid of stuff we really don't need - maybe put it in %doc # %{__rm} -f %{buildroot}%{_localstatedir}/lib/asterisk/moh/.asterisk-moh-freeplay-wav 2>/dev/null %{__rm} -f %{buildroot}%{_localstatedir}/lib/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.4 2>/dev/null %{__rm} -f %{buildroot}%{_localstatedir}/lib/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.6 2>/dev/null %{__rm} -f %{buildroot}%{_localstatedir}/lib/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.7 2>/dev/null # %{__rm} -rf %{buildroot}%{_localstatedir}/lib/asterisk/static-http 2>/dev/null # %{__rm} -f %{buildroot}%{_localstatedir}/lib/asterisk/agi-bin/* 2>/dev/null %{__rm} -f %{buildroot}%{_localstatedir}/lib/asterisk/sounds/.asterisk-core-sounds-en-gsm-1.4.8 %pre %{_sbindir}/groupadd -g 330 -r %{prename} &>/dev/null || : %{_sbindir}/useradd -u 330 -r -s /sbin/false -d /var/lib/asterisk -M \ -c 'Asterisk PBX' -g asterisk asterisk &>/dev/null || : %post /sbin/chkconfig --add asterisk mkdir -p -m 0775 /var/run/asterisk chown asterisk:asterisk /var/run/asterisk chown asterisk:asterisk /var/log/asterisk mv /etc/asterisk/manager.conf /etc/asterisk/manager.conf.bak && cat /etc/asterisk/manager.conf.bak | grep -v 'webenabled' | sed -e '/\[general\]/awebenabled = yes' > /etc/asterisk/manager.conf %preun if [ $1 -eq 0 ]; then /sbin/service asterisk stop &> /dev/null || : /sbin/chkconfig --del asterisk fi %clean rm -rf $RPM_BUILD_ROOT %files %defattr(-,root,root,-) %doc BUGS CHANGES ChangeLog COPYING CREDITS doc/* configs LICENSE sample.call README* *.txt %doc static-http agi/*-test agi/*.agi %{_initrddir}/asterisk %dir %{_libdir}/asterisk %dir %{_libdir}/asterisk/modules %{_libdir}/asterisk/modules/* %{_localstatedir}/lib/asterisk/moh/* %{_localstatedir}/lib/asterisk/static-http/* %{_localstatedir}/lib/asterisk/agi-bin/* %{_localstatedir}/lib/asterisk/firmware/* %{_localstatedir}/lib/asterisk/images/* %{_localstatedir}/lib/asterisk/keys/* %{_localstatedir}/lib/asterisk/sounds/* %{_localstatedir}/lib/asterisk/sounds/.* %{_sbindir}/aelparse %{_sbindir}/asterisk %{_sbindir}/astgenkey %{_sbindir}/astman %{_sbindir}/autosupport %{_sbindir}/muted %{_sbindir}/rasterisk %{_sbindir}/safe_asterisk %{_sbindir}/smsq %{_sbindir}/stereorize %{_sbindir}/streamplayer %{_mandir}/man8/asterisk.8* %{_mandir}/man8/astgenkey.8* %{_mandir}/man8/autosupport.8* %{_mandir}/man8/safe_asterisk.8* /var/spool/asterisk/voicemail/* %dir %{_sysconfdir}/asterisk %config(noreplace) %{_sysconfdir}/asterisk/* %files alsa %defattr(-,root,root,-) %{_libdir}/asterisk/modules/*alsa* %config(noreplace)%attr(640,asterisk,root)/etc/asterisk/alsa.conf %if %build_h323 == 1 %files h323 %defattr(-,root,root,-) %{_libdir}/asterisk/modules/*h323* %config(noreplace)%attr(640,asterisk,root)/etc/asterisk/h323.conf %endif %files odbc %defattr(-,root,root,-) %{_libdir}/asterisk/modules/*odbc* %config(noreplace)%attr(640,asterisk,root)/etc/asterisk/*odbc* %if ! 0%{?suse_version} == 1010 %files pgsql %defattr(-,root,root,-) %{_libdir}/asterisk/modules/*pgsql* %config(noreplace)%attr(640,asterisk,root)/etc/asterisk/*pgsql* %endif %files spandsp %defattr(-,root,root,-) %{_libdir}/asterisk/modules/*fax* %files zaptel %defattr(-,root,root,-) %{_libdir}/asterisk/modules/*zap* %config(noreplace)%attr(640,asterisk,root)/etc/asterisk/zapata.conf %files devel %defattr(-,root,root,-) %dir %{_includedir}/asterisk %{_includedir}/asterisk.h %{_includedir}/asterisk/*.h %changelog -n asterisk * Wed May 21 2008 Carsten Schoene <cs@linux-administrator.com> - version bump to 1.4.20 * Wed May 13 2008 Carsten Schoene <cs@linux-administrator.com> - version bump to 1.4.19.2 * Wed Apr 23 2008 Carsten Schoene <cs@linux-administrator.com> - version bump to 1.4.19.1 * Sat Apr 05 2008 Carsten Schoene <cs@linux-administrator.com> - version bump to 1.4.19 * Wed Mar 19 2008 Carsten Schoene <cs@linux-administrator.com> - version bump to 1.4.18.1 * Sat Feb 09 2008 Carsten Schoene <cs@linux-administrator.com> - version bump to 1.4.18 * Fri Jan 04 2008 - vittorio@vitsoft.bz - Modified for asterisk 1.4.17 - Security fix for chan_sip * Tue Nov 27 2007 - vittorio@vitsoft.bz - Modified for asterisk 1.4.15 * Thu Oct 04 2007 - vittorio@vitsoft.bz - Modified for asterisk 1.4.12 - removed bristuff and added misdn support - added res_config_ldap for realtime LDAP * Mon Aug 13 2007 - vittorio@vitsoft.bz - Modified for asterisk 1.4.10 * Thu Nov 16 2006 - max@suse.de - Removed the mp3 files from the tarball and added a check to the spec file to prevent forgetting it again at the next update. - Fixed app_dtmftotext to work with the latest spandsp. * Fri Oct 20 2006 - max@suse.de - New versions: asterisk-1.2.13, bristuff-0.3.0-PRE-1s. - Fixes a heap overflow in chan_skinny (#213579). * Fri Jul 14 2006 - max@suse.de - New versions: asterisk-1.2.9.1, bristuff-0.3.0-PRE-1q: - Fixes a buffer overflow in chan_iax.c (#182333, CVE-2006-2898). - Many more enhancements and bug fixes. - Use the libgsm package instead of builtin copy (#102354, asterisk-gsm.patch). * Thu Apr 27 2006 - max@suse.de - Removed example mp3 files from the source tarball. This makes the source and binary package considerably smaller. (Bug #118226) * Mon Apr 24 2006 - max@suse.de - Fixed a buffer overflow in format_jpeg.c. (Bug #168274, CVE-2006-1827) * Mon Mar 20 2006 - max@suse.de - New versions: asterisk-1.2.5, bristuff-0.3.0-PRE-1k: - Makefile: Bug 6638 - Use POSIX command for Solaris - build_tools/make_build_h: Bug 6638 - Change from a historic BSD command to a POSIX command for determining username - asterisk.c: Bug 6637 - Fixes for Solaris - Makefile: If debugging, the frame pointer is helpful - res/res_monitor.c: fix inaccurate ack message to ChangeMonitor action (issue #6630) - asterisk.sgml: make the terminology used in the synopsis match the option description - asterisk.sgml: add the -L option to the synopsis on the man page - cdr/cdr_manager.c, cdr/cdr_tds.c, res/res_config_odbc.c, include/asterisk/cdr.h, cdr/cdr_pgsql.c, cdr/cdr_odbc.c, cdr.c: Bug 6615 - Fix 64bit conversion errors by using a long int - build_tools/make_svn_branch_name: Bug 6618 - Solaris compatibility fix - channels/chan_iax2.c: fix the output that indicates whether qualify smoothing is on or not (issue #6608) - asterisk.c: adjust the keys directory when astvarlibdir is specified in asterisk.conf (issue #6602) - res/res_agi.c: add a missing newline in the agi app description - cli.c: don't try to print the help text for a CLI command when RESULT_SHOWUSAGE is returned if there is no help text available (issue #6604) - channels/chan_sip.c: fix finding realtime peers that are not dynamic by ip address (issue #6093) - channel.c: don't hang up the channel if its state is set to UP before we return from ast_call (issue #6569) - include/asterisk/logger.h, logger.c: Bug 5950 - reenable queue log rotation; also, eliminate redundant code - translate.c: Backport of fix to translation optimizations. - translate.c: factor the number of translation steps required into translation path decisions, so that equal cost paths that require fewer translations are preferred - translate.c: reformat code to fit guidelines remember which translation paths are multi-step paths - channel.c: ensure that spy frame queueing is able to deal with translation failing for any reason (issue #6546) - Makefile: set PWD properly - dnsmgr.c, include/asterisk/linkedlists.h: backport list handling fix from trunk (solves memory leak problem in cdr variables and device state watchers) remove unused variable to silence compiler warning - configs/iax.conf.sample: add comment warning people about trying to use hostnames/IPs in the sample config - app.c: Would be nice to tell people to look in the right file to increase a constant - channels/chan_sip.c: Handle ACKing properly (remove gratuitous -1) - channels/chan_iax2.c: Fix numerous places in jitter buffer where freed memory is referenced - formats/format_sln.c: Okay, fseek doesn't return an offset - apps/app_voicemail.c: Fix possible lack of initialization of useadsi - formats/format_sln.c: Bug 6539 - Division by two negates error flag - app.c: Bug 6529 - memory leak in ast_play_and_prepend - jitterbuf.c: fix incorrent index calculation for jitterbuffer history (issue #6517) - apps/app_voicemail.c: when executing the Directory application from voicemail and a context is not specified, use the "default" context, not the channel's current context (issue #6507) - channels/chan_agent.c: ensure that agents logged in via the manager interface are stored in the persistence database (related to issue #6301) - funcs/func_enum.c: handle longer ENUM lookup results (issue #6476) - res/res_agi.c: ensure that FastAGI launcher can handle system call interruption (issue #6449) - apps/app_meetme.c: bug fix from 6485 with musiconhold not being turned off by app_meetme - apps/app_queue.c: don't double-increment abandon counter for calls that are hung up while dialing members (issue #6289) - apps/app_meetme.c: Fix stopstream in menus (bug #6137) - asterisk.c: #ifdef the include too. - asterisk.c: #ifdef'd the prctl fix to only try and compile on linux systems. Thanks rizzo for pointing this out. - channels/chan_sip.c: when answering INVITE, don't send codecs the peer didn't offer (issue #6052) - rtp.c: revert yesterday's temporary fix for issue #6052 - asterisk.c: Fixed my silly backport error from r9861 - asterisk.c: Merged changes from r9844 from /trunk. Make sure that PR_SET_DUMPABLE is set to make certain that we still dump core if Asterisk has setuid'd to run as non-root. - rtp.c: don't try to use peer's dynamic codec numbers, it leads to duplication (issue #6052) - apps/app_meetme.c: Don't set the formats before we stop indications. (issue #6380) - channels/chan_mgcp.c, channels/chan_sip.c, pbx/pbx_dundi.c, channels/chan_iax2.c: fix memory leak from not destroying the scheduler context on module unload - apps/app_page.c: fix due to CDR changes - manager.c, pbx/pbx_spool.c, include/asterisk/channel.h, include/asterisk/pbx.h, include/asterisk/manager.h, channel.c, pbx.c: now that CDR is a loadable module, don't depend on it elsewhere (issue #6460) - channels/chan_sip.c, cdr.c: clean up my mess from thread-starting change - channels/chan_sip.c: kpfleming's fix from r9472 backported to 1.2 - channels/chan_mgcp.c, dnsmgr.c, channels/chan_sip.c, devicestate.c, channels/chan_modem.c, cdr.c: don't create monitor threads in detached mode, when we need to be able to pthread_join() them later if the module is unloaded (solve crash-on-unload problem for these channel modules) - apps/app_voicemail.c: Revert behavior change from previous commit (fixes only) - apps/app_voicemail.c: Backport 5929 to 1.2 - apps/Makefile: add another location for postgresql headers (issue [#6419]) - channels/chan_iax2.c: reload peercontext on iax2 reload (issue #6442) - cdr/Makefile: Leave it to RH/CentOS to put the freetds headers in a completely nonstandard location. - logger.c, channels/chan_oss.c: Make logger report error,warning,notice if logger.conf not found, also updated chan_oss to give correct error message if its config file is not found. - apps/app_macro.c: Bug 6176 - Fix race condition - Makefile: don't override ASTERISKVERSIONNUM to 000000 for non-svn builds - res/res_odbc.c: Fix for (#6309), potential (highly unlikely) memory leak in res_odbc - channels/chan_zap.c: disable buggy PRI user-user code until it can be fixed - channels/chan_sip.c: Issue 6182 - Don't remove scheduled event until it's really done. (reported by malverian) - channels/chan_sip.c: Issue 6362 - Register without Contact: and Expires: fails (reporter: op) - ast_expr2.h, ast_expr2f.c, ast_expr2.c: Bug 6072 - Revisions to the source bison and flex files don't auto-regenerate these files - channels/chan_zap.c: fix problem with dtmf on e&m (issue #6364) - channels/chan_sip.c: Issue 5898: Registrations does not get deleted if there's an active SIP dialog - channels/chan_sip.c: don't call ast_update_realtime with uninitialized variables if we get a registration with an expirey of 0 seconds (issue #6173) - channels/chan_features.c: fix memory leak (inspired by issue #6351) * Wed Feb 15 2006 - max@suse.de - Remove unwanted capi.conf . * Fri Feb 10 2006 - max@suse.de - Makefile, keep your hands off -march (#148593). * Mon Feb 06 2006 - max@suse.de - Included all subdirs of /var/log/asterisk in file list. * Mon Jan 30 2006 - mls@suse.de - converted neededforbuild to BuildRequires * Wed Jan 25 2006 - max@suse.de - New versions: asterisk-1.2.3, bristuff-0.3.0-PRE-1i . * Mon Jan 23 2006 - max@suse.de - New version of the spandsp patch: 0.0.2pre23 * Tue Jan 17 2006 - max@suse.de - Don't use -fsigned-char on PPC (bug #93872). * Tue Dec 20 2005 - max@suse.de - New version: 1.2.1 - New version of the bristuff patch: 0.3.0-PRE-1c - New version of the spandsp patch: 0.0.2pre21 * Thu Aug 11 2005 - max@suse.de - New version of the bristuff patch: 0.2.0-rc8n * Tue Jul 19 2005 - max@suse.de - Disabled the mmx assembler stuff in gsm again. * Mon Jul 18 2005 - max@suse.de - New version 1.0.9 - New version of the bristuff patch: 0.2.0-rc8j * Thu Apr 14 2005 - pth@suse.de - remove a bogus non-static declaration that conflicts with a static one. * Mon Mar 07 2005 - max@suse.de - New version of the bristuff patch: 0.2.0-rc7k * Mon Feb 28 2005 - max@suse.de - New version: 1.0.6. - New version of the bristuff patch: 0.2.0-rc7i - New version of the spandsp patch: 0.0.2pre10 * Thu Jan 13 2005 - max@suse.de - New version: 1.0.3. - New version of the bristuff patch: 0.2.0-rc3a - New version of the spandsp patch: 0.0.2pre8 * Fri Nov 26 2004 - max@suse.de - Update spandsp patch to version 0.0.2pre6 * Fri Nov 05 2004 - max@suse.de - New version: 1.0.2 - New version of the bristuff patch: 0.2.0-rc2 - New version of the spandsp patch: 0.0.2pre4 * Tue Oct 05 2004 - max@suse.de - Fixed location of pid file in init script [Bug #46859]. - Added the /var/log/asterisk/cdr-csv directory to the package. * Wed Sep 29 2004 - max@suse.de - Added /usr/sbin/useradd to PreReq [Bug #46487]. * Wed Sep 29 2004 - max@suse.de - /var/lib/asterisk also needs to be owned by the asterisk user. * Mon Sep 20 2004 - max@suse.de - Creating a user named asterisk during installation under which the asterisk server will be run and who will own the config files and asterisk's /var/log, /var/spool, and /var/run areas. - Added asterisk as a prerequirement to all subpackages to make sure the user exists when the package is installed. * Mon Sep 13 2004 - ro@suse.de - Fix usage of __P (patch from db1 package) * Mon Sep 06 2004 - max@suse.de - Created a separate source package for asterisk-capi. * Fri Aug 20 2004 - max@suse.de - Update bristuff patch to version 0.1.0-RC4a - Update to CVS snapshot from 2004-08-14. - Use platform independent names for linking against libpt and libopenh323. - Temporarily disablied building of H.323 support due to incompatible changes in openh323. * Mon Aug 16 2004 - ro@suse.de - try to fix build on x86_64 and ia64 * Fri Aug 13 2004 - max@suse.de - Update asterisk to CVS snapshot of 2004-08-12 - Enable H.323 support - Put modules with external library dependencies into subpackages - Update bristuff patch to version 0.1.0-RC3 * Wed Jul 28 2004 - max@suse.de - Fixed db on ppc64. - Removed some more bogus -march switches. - Disables loading of the CAPI modules by default. - Added init script. * Fri Jul 23 2004 - max@suse.de - Update asterisk to CVS snapshot from 2004-07-22 - Update chan_capi to version 0.3.4b - Update bristuff patch to version 0.1.0-RC2 * Thu Jul 08 2004 - max@suse.de - Don't try to be clever on detecting optimization, rpmbuild provides us with the necessary flags. - Don't build chan_capi on s390/s390x, because there is no CAPI support on these platforms. * Wed Jun 23 2004 - max@suse.de - New package: Asterisk (CVS snapshot from 2004-10-05). Asterisk a complete PBX in software (www.asterisk.org). - Added chan_capi for using CAPI devices as Asterisk channels. - Added patch to support softfax using the spandsp library. - Added bristuff patches to support BRI ISDN cards using libpri.
