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Bump PKGREVISION for perl update.
2011-08-07 04:40:32 +02:00
# $NetBSD: Makefile,v 1.72 2011/08/07 02:40:32 jnemeth Exp $
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
Update to 1.2.37. This update is to fix two security issues. 1.2.36 fixed AST-2009-008, and 1.2.37 fixed AST-2009-010. The problem in AST-2009-008 is: ----- It is possible to determine if a peer with a specific name is configured in Asterisk by sending a specially crafted REGISTER message twice. The username that is to be checked is put in the user portion of the URI in the To header. A bogus non-matching value is put into the username portion of the Digest in the Authorization header. If the peer does exist the second REGISTER will receive a response of "403 Authentication user name does not match account name". If the peer does not exist the response will be "404 Not Found" if alwaysauthreject is disabled and "401 Unauthorized" if alwaysauthreject is enabled. ----- And, the problem in AST-2009-010 is: ----- An attacker sending a valid RTP comfort noise payload containing a data length of 24 bytes or greater can remotely crash Asterisk. -----
2009-12-18 15:39:26 +01:00
DISTNAME= asterisk-1.2.37
Bump PKGREVISION for perl update.
2011-08-07 04:40:32 +02:00
PKGREVISION= 3
Changes 1.0.9: -- fix bug in callerid matching in the dialplan that was introduced in 1.0.8 Changes 1.0.8: -- chan_zap -- Asterisk will now also look in the regular context for the fax extension while executing a macro. Previously, for this to work, the fax extension would have to be included in the macro definition. -- On some systems, ALERTING will be sent after PROCEEDING, so code has been added to account for this case. -- If no extension is specified on an overlap call, the 's' extension will be used. -- chan_sip -- We no longer send a "to" tag on "100 Trying" messages, as it is inappropriate to do so. -- We now respond correctly to an invite for T.38 with a "488 Not acceptable here" -- We now discard saved tags on 401/407 responses in case the provider we're talking to tries to pull a dirty trick on us and change it. -- rtptimeout options will now be correctly set on a peer basis rather than only global -- chan_mgcp -- Fixed setting of accountcode -- Fixed where *67 to block callerid only worked for first call -- chan_agent -- We now will not pass audio until the agent has acked the call if the configuration is set up for the agent to do so. -- chan_alsa -- Fixed problems with the unloading of this module -- res_agi -- A fix has been added to prevent calls from being hung up when more than one call is executing an AGI script calling the GET DATA command. -- AGI scripts will now continue to run even if a file was not found with the GET DATA command. -- When calling SAY NUMBER with a number like 09, we will now say "nine" instead of "zero" -- app_dial -- There was a problem where text frames would not be forwarded before the channel has been answered. -- app_disa -- Fixed the timeout used when no password is set -- app_queue -- Distinctive ring has been fixed to work for queue members -- rtp -- Fixed a logic error when setting the "rtpchecksums" option -- say.c -- A problem has been fixed with saying the date in Spanish. -- Makefile -- A line was missing for the autosupport script that caused "make rpm" to fail -- format_wav_gsm -- Fixed a problem with wav formatting that prevented files from being played in some media players -- pbx_spool -- Fixed if the last line of text in a file for the call spool did not contain a new line, it would not be processed -- logger -- Fixed the logger so that color escape sequences wouldn't be sent to the logs -- format_sln -- A lot of changes were made to correctly handle signed linear format on big endian machines
2005-09-02 14:58:34 +02:00
CATEGORIES= comms net audio
new MASTER_SITES
2009-05-15 20:24:29 +02:00
MASTER_SITES= http://downloads.asterisk.org/pub/telephony/asterisk/releases/ \
http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
pkgsrc-users, not packages (hi riz!)
2008-06-18 13:12:53 +02:00
MAINTAINER= pkgsrc-users@NetBSD.org
Changes 1.0.9: -- fix bug in callerid matching in the dialplan that was introduced in 1.0.8 Changes 1.0.8: -- chan_zap -- Asterisk will now also look in the regular context for the fax extension while executing a macro. Previously, for this to work, the fax extension would have to be included in the macro definition. -- On some systems, ALERTING will be sent after PROCEEDING, so code has been added to account for this case. -- If no extension is specified on an overlap call, the 's' extension will be used. -- chan_sip -- We no longer send a "to" tag on "100 Trying" messages, as it is inappropriate to do so. -- We now respond correctly to an invite for T.38 with a "488 Not acceptable here" -- We now discard saved tags on 401/407 responses in case the provider we're talking to tries to pull a dirty trick on us and change it. -- rtptimeout options will now be correctly set on a peer basis rather than only global -- chan_mgcp -- Fixed setting of accountcode -- Fixed where *67 to block callerid only worked for first call -- chan_agent -- We now will not pass audio until the agent has acked the call if the configuration is set up for the agent to do so. -- chan_alsa -- Fixed problems with the unloading of this module -- res_agi -- A fix has been added to prevent calls from being hung up when more than one call is executing an AGI script calling the GET DATA command. -- AGI scripts will now continue to run even if a file was not found with the GET DATA command. -- When calling SAY NUMBER with a number like 09, we will now say "nine" instead of "zero" -- app_dial -- There was a problem where text frames would not be forwarded before the channel has been answered. -- app_disa -- Fixed the timeout used when no password is set -- app_queue -- Distinctive ring has been fixed to work for queue members -- rtp -- Fixed a logic error when setting the "rtpchecksums" option -- say.c -- A problem has been fixed with saying the date in Spanish. -- Makefile -- A line was missing for the autosupport script that caused "make rpm" to fail -- format_wav_gsm -- Fixed a problem with wav formatting that prevented files from being played in some media players -- pbx_spool -- Fixed if the last line of text in a file for the call spool did not contain a new line, it would not be processed -- logger -- Fixed the logger so that color escape sequences wouldn't be sent to the logs -- format_sln -- A lot of changes were made to correctly handle signed linear format on big endian machines
2005-09-02 14:58:34 +02:00
HOMEPAGE= http://www.asterisk.org/
COMMENT= The Asterisk Software PBX
Digium in its infinite wisdom changed the Music-On-Hold sound files in all release tarballs. Update for that change. While here, do some pkglint cleanup and add LICENSE=gplv2.
2009-08-21 00:31:41 +02:00
LICENSE= gnu-gpl-v2
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
Add DESTDIR support.
2008-06-12 04:14:13 +02:00
PKG_DESTDIR_SUPPORT= user-destdir
Fixed the remaining pkglint -Wall warnings.
2006-01-29 02:30:20 +01:00
.include "../../mk/bsd.prefs.mk"
Add bash to tools for mkpkgconfig.
2008-02-27 13:31:12 +01:00
USE_TOOLS+= bash bison gmake perl:run
Use REPLACE_BASH to make sure right bash is found for mkpkgconfig.
2008-02-28 09:53:31 +01:00
REPLACE_BASH= mkpkgconfig
- Fixed some pkglint warnings. - Replaced absolute directories like /usr/pkg and /var with ${PREFIX} and ${VARBASE}. - USE_TOOLS+=perl:run, since there is one Perl program installed with the package. - Bumped PKGREVISION.
2006-01-29 02:21:45 +01:00
REPLACE_PERL+= agi/agi-test.agi
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
Fixed the remaining pkglint -Wall warnings.
2006-01-29 02:30:20 +01:00
INSTALL_TARGET= install samples
Add INSTALLATION_DIRS so that installation is successful even in a bulk build.
2008-05-26 14:29:24 +02:00
INSTALLATION_DIRS= lib/pkgconfig
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
- make sure rc.d script can find asterisk when it isn't in the path - pkglint
2008-11-24 10:27:29 +01:00
BUILD_DEFS+= VARBASE
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
# Running as non-root will come later
#ASTERISK_USER?= asterisk
#ASTERISK_GROUP?= asterisk
#PKG_GROUPS= ${ASTERISK_GROUP}
Modify packages that set PKG_USERS and PKG_GROUPS to follow the new syntax as specified in pkgsrc/mk/install/bsd.pkginstall.mk:1.47.
2006-04-23 02:12:35 +02:00
#PKG_USERS= ${ASTERISK_USER}:${ASTERISK_GROUP}
Make it easier to build and install packages "unprivileged", where the owner of all installed files is a non-root user. This change affects most packages that require special users or groups by making them use the specified unprivileged user and group instead. (1) Add two new variables PKG_GROUPS_VARS and PKG_USERS_VARS to unprivileged.mk. These two variables are lists of other bmake variables that define package-specific users and groups. Packages that have user-settable variables for users and groups, e.g. apache and APACHE_{USER,GROUP}, courier-mta and COURIER_{USER,GROUP}, etc., should list these variables in PKG_USERS_VARS and PKG_GROUPS_VARS so that unprivileged.mk can know to set them to ${UNPRIVILEGED_USER} and ${UNPRIVILEGED_GROUP}. (2) Modify packages to use PKG_GROUPS_VARS and PKG_USERS_VARS.
2007-07-04 22:54:31 +02:00
#PKG_GROUPS_VARS+= ASTERISK_GROUP
#PKG_USERS_VARS+= ASTERISK_USER
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
# Various path settings for Asterisk
Changes 1.2.8: * Number of bug fixes, including IAX2 channel driver fixes.
2006-05-31 20:43:15 +02:00
PKG_SYSCONFSUBDIR= asterisk
- Fixed some pkglint warnings. - Replaced absolute directories like /usr/pkg and /var with ${PREFIX} and ${VARBASE}. - USE_TOOLS+=perl:run, since there is one Perl program installed with the package. - Bumped PKGREVISION.
2006-01-29 02:21:45 +01:00
ASTLIBDIR?= ${PREFIX}/lib/asterisk
ASTVARLIBDIR?= ${PREFIX}/libdata/asterisk
ASTETCDIR?= ${PKG_SYSCONFDIR}
ASTSPOOLDIR?= ${VARBASE}/spool/asterisk
ASTLOGDIR?= ${VARBASE}/log/asterisk
ASTHEADERDIR?= ${PREFIX}/include/asterisk
ASTCONFPATH?= ${ASTETCDIR}/asterisk.conf
ASTBINDIR?= ${PREFIX}/bin
ASTSBINDIR?= ${PREFIX}/sbin
ASTVARRUNDIR?= ${VARBASE}/run
Update asterisk to 1.2.31. While here, update MASTER_SITES and honor PKGMANDIR. ChangeLog-1.2.31: 2009-01-06 Leif Madsen <lmadsen@digium.com> * Asterisk 1.2.31 released 2009-01-06 20:44 +0000 [r167259] Tilghman Lesher <tlesher@digium.com> * channels/chan_iax2.c: Security fix AST-2009-001. 2008-12-10 Tilghman Lesher <tlesher@digium.com> * Asterisk 1.2.30.4 released 2008-12-10 21:06 +0000 [r162868] Tilghman Lesher <tlesher@digium.com> * channels/chan_iax2.c: Fix for AST-2008-012 2008-12-05 20:50 +0000 [r161421] Sean Bright <sean.bright@gmail.com> * include/asterisk/astobj2.h, astobj2.c: Fix build errors on FreeBSD (uint -> unsigned int). (closes issue #14006) Reported by: alphaque Patches: astobj2.h-patch uploaded by alphaque (license 259) (Slightly modified by seanbright) 2008-12-01 Tilghman Lesher <tlesher@digium.com> * Asterisk 1.2.30.3 released 2008-11-25 21:37 +0000 [r159245] Tilghman Lesher <tlesher@digium.com> * channels/chan_iax2.c: Regression fix for last security fix. Set the iseqno correctly. (closes issue #13918) Reported by: ffloimair Patches: 20081119__bug13918.diff.txt uploaded by Corydon76 (license 14) Tested by: ffloimair 2008-08-09 Tilghman Lesher <tlesher@digium.com> * Asterisk 1.2.30.2 released 2008-08-09 15:24 +0000 [r136945] Tilghman Lesher <tlesher@digium.com> * include/asterisk/compat.h, include/asterisk/astobj2.h: Regression fixes for Solaris 2008-07-25 15:00 +0000 [r133577] Russell Bryant <russell@digium.com> * LICENSE: Fix the IAX2 URI for calling Digium 2008-07-23 Tilghman Lesher <tlesher@digium.com> * Asterisk 1.2.30.1 released 2008-07-24 03:46 +0000 [r133360] Tilghman Lesher <tlesher@digium.com> * channels/chan_iax2.c: This part was not correctly patched for AST-2008-010.
2009-01-21 06:35:07 +01:00
ASTMANDIR?= ${PREFIX}/${PKGMANDIR}
- Fixed some pkglint warnings. - Replaced absolute directories like /usr/pkg and /var with ${PREFIX} and ${VARBASE}. - USE_TOOLS+=perl:run, since there is one Perl program installed with the package. - Bumped PKGREVISION.
2006-01-29 02:21:45 +01:00
ASTEXAMPLEDIR?= ${PREFIX}/share/examples/asterisk
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
# Override default paths in Makefile
Fixed the remaining pkglint -Wall warnings.
2006-01-29 02:30:20 +01:00
SUBST_CLASSES+= asterisk
SUBST_STAGE.asterisk= pre-configure
Patch the default path for music-on-hold files.
2006-01-24 03:09:38 +01:00
SUBST_FILES.asterisk= Makefile configs/musiconhold.conf.sample
Changes 1.2.7: * Important bug fixes * SIP handling * MixMonitor call recording
2006-04-13 10:47:06 +02:00
SUBST_SED.asterisk+= -e 's|/var/lib/asterisk|${ASTVARLIBDIR}|'
SUBST_SED.asterisk+= -e 's|^ *\(ASTLIBDIR\)=.*|\1 = ${ASTLIBDIR}|'
SUBST_SED.asterisk+= -e 's|^ *\(ASTVARLIBDIR\)=.*|\1 = ${ASTVARLIBDIR}|'
SUBST_SED.asterisk+= -e 's|^ *\(ASTETCDIR\)=.*|\1 = ${ASTETCDIR}|'
SUBST_SED.asterisk+= -e 's|^ *\(ASTSPOOLDIR\)=.*|\1 = ${ASTSPOOLDIR}|'
SUBST_SED.asterisk+= -e 's|^ *\(ASTLOGDIR\)=.*|\1 = ${ASTLOGDIR}|'
SUBST_SED.asterisk+= -e 's|^ *\(ASTHEADERDIR\)=.*|\1 = ${ASTHEADERDIR}|'
- make sure rc.d script can find asterisk when it isn't in the path - pkglint
2008-11-24 10:27:29 +01:00
SUBST_SED.asterisk+= -e 's|^ *\(ASTCONFPATH\)=.*|\1 = ${ASTCONFPATH:Q}|'
Changes 1.2.7: * Important bug fixes * SIP handling * MixMonitor call recording
2006-04-13 10:47:06 +02:00
SUBST_SED.asterisk+= -e 's|^ *\(ASTBINDIR\)=.*|\1 = ${ASTBINDIR}|'
SUBST_SED.asterisk+= -e 's|^ *\(ASTSBINDIR\)=.*|\1 = ${ASTSBINDIR}|'
SUBST_SED.asterisk+= -e 's|^ *\(ASTVARRUNDIR\)=.*|\1 = ${ASTVARRUNDIR}|'
SUBST_SED.asterisk+= -e 's|^ *\(ASTMANDIR\)=.*|\1 = ${ASTMANDIR}|'
SUBST_SED.asterisk+= -e 's|^ *\(ASTEXAMPLEDIR\)=.*|\1 = ${ASTEXAMPLEDIR}|'
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
.include "options.mk"
Fixed the remaining pkglint -Wall warnings.
2006-01-29 02:30:20 +01:00
RCD_SCRIPTS= asterisk
OWN_DIRS+= ${ASTSPOOLDIR} ${ASTLOGDIR} ${ASTVARRUNDIR}
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
OWN_DIRS+= ${ASTLOGDIR}/cdr-csv
Need to care ${ASTVARLIBDIR}/sounds/priv-callerintros. XXX: it should be in ${VARBASE}, not ${PREFIX}/libdata.
2009-01-22 13:19:49 +01:00
# XXX: should be in ${VARBASE}
OWN_DIRS+= ${ASTVARLIBDIR}/sounds/priv-callerintros
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
CONF_FILES= # empty
- Fixed some pkglint warnings. - Replaced absolute directories like /usr/pkg and /var with ${PREFIX} and ${VARBASE}. - USE_TOOLS+=perl:run, since there is one Perl program installed with the package. - Bumped PKGREVISION.
2006-01-29 02:21:45 +01:00
.for f in asterisk.conf extensions.conf
Digium in its infinite wisdom changed the Music-On-Hold sound files in all release tarballs. Update for that change. While here, do some pkglint cleanup and add LICENSE=gplv2.
2009-08-21 00:31:41 +02:00
CONF_FILES+= ${ASTEXAMPLEDIR}/${f:Q} ${ASTETCDIR}/${f:Q}
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
.endfor
Digium in its infinite wisdom changed the Music-On-Hold sound files in all release tarballs. Update for that change. While here, do some pkglint cleanup and add LICENSE=gplv2.
2009-08-21 00:31:41 +02:00
MAKE_ENV+= VARBASE=${VARBASE}
MAKE_FLAGS+= INSTALL_PREFIX=${PREFIX}
- Fixed some pkglint warnings. - Replaced absolute directories like /usr/pkg and /var with ${PREFIX} and ${VARBASE}. - USE_TOOLS+=perl:run, since there is one Perl program installed with the package. - Bumped PKGREVISION.
2006-01-29 02:21:45 +01:00
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
# if we put all the files in $CONF_FILES, the message is _way_ too long.
- Fixed some pkglint warnings. - Replaced absolute directories like /usr/pkg and /var with ${PREFIX} and ${VARBASE}. - USE_TOOLS+=perl:run, since there is one Perl program installed with the package. - Bumped PKGREVISION.
2006-01-29 02:21:45 +01:00
.for f in adsi.conf adtranvofr.conf agents.conf alarmreceiver.conf \
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
alsa.conf cdr_manager.conf cdr_odbc.conf \
cdr_pgsql.conf cdr_tds.conf enum.conf extconfig.conf \
features.conf festival.conf iax.conf \
iaxprov.conf indications.conf logger.conf manager.conf \
meetme.conf mgcp.conf modem.conf modules.conf musiconhold.conf \
osp.conf oss.conf phone.conf privacy.conf queues.conf \
Update to asterisk 1.2.3 - bugfixes only.
2006-01-25 18:48:43 +01:00
res_odbc.conf rpt.conf rtp.conf sip.conf \
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
skinny.conf voicemail.conf vpb.conf zapata.conf asterisk.adsi \
telcordia-1.adsi
Digium in its infinite wisdom changed the Music-On-Hold sound files in all release tarballs. Update for that change. While here, do some pkglint cleanup and add LICENSE=gplv2.
2009-08-21 00:31:41 +02:00
CONF_FILES+= ${ASTEXAMPLEDIR}/${f:Q} ${ASTETCDIR}/${f:Q}
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
.endfor
Asterisk doesn't work with non-native threads. Approved by jmcneill.
2005-04-13 21:22:17 +02:00
PTHREAD_OPTS+= require native
Make this compile with newer versions of Darwin that have poll(2).
2005-10-10 23:45:08 +02:00
.if (${OPSYS} == "Darwin" && exists(/usr/include/sys/poll.h))
post-patch:
Digium in its infinite wisdom changed the Music-On-Hold sound files in all release tarballs. Update for that change. While here, do some pkglint cleanup and add LICENSE=gplv2.
2009-08-21 00:31:41 +02:00
${ECHO} "#include <sys/poll.h>" > ${WRKSRC}/include/asterisk/poll-compat.h
Make this compile with newer versions of Darwin that have poll(2).
2005-10-10 23:45:08 +02:00
.endif
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
.include "../../devel/ncurses/buildlink3.mk"
This package needs zlib to build.
2006-06-21 01:32:36 +02:00
.include "../../devel/zlib/buildlink3.mk"
This package needs openssl to build; include its buildlink3 file.
2005-04-17 05:48:59 +02:00
.include "../../security/openssl/buildlink3.mk"
PR/38351 - Miro Voutilainen -- app_curl does not build
2009-01-26 14:15:49 +01:00
.include "../../www/curl/buildlink3.mk"
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
.include "../../mk/pthread.buildlink3.mk"
.include "../../mk/bsd.pkg.mk"
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