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Update asterisk to 1.2.23 * channels/chan_iax2.c: Don't create the Asterisk channel until we are starting the PBX on it. (ASA-2007-018) * channels/chan_agent.c: (closes issue #5866) Reported by: tyler Do not force channel format changes when a generator is present. The generator may have changed the formats itself and changing them back would cause issues. * channels/chan_sip.c: (closes issue #10236) Reported by: homesick Patches: rpid_1.4_75840.patch uploaded by homesick (license 91) Accept Remote Party ID on guest calls. * include/asterisk/app.h: We should not use C++ reserved words in API headers (closes issue #10266) * channels/chan_sip.c: Backport a fix for a memory leak that was fixed in trunk in reivision 76221 by rizzo. The memory used for the localaddr list was not freed during a configuration reload. * channels/chan_sip.c: (closes issue #10247) Reported by: fkasumovic Patches: chan_sip.patch uploaded by fkasumovic (license #101) Drop any peer realm authentication entries when reloading so multiple entries do not get added to the peer. * channels/chan_iax2.c: When processing full frames, take sequence number wraparound into account when deciding whether or not we need to request retransmissions by sending a VNAK. This code could cause VNAKs to be sent erroneously in some cases, and to not be sent in other cases when it should have been. (closes issue #10237, reported and patched by mihai) * channels/chan_iax2.c: When traversing the queue of frames for possible retransmission after receiving a VNAK, handle sequence number wraparound so that all frames that should be retransmitted actually do get retransmitted. (issue #10227, reported and patched by mihai) * apps/app_voicemail.c: Store prior to copy (closes issue #10193) * apps/app_queue.c: removed the word 'pissed' from ast_log(...)
2007-08-04 00:40:00 +02:00
# $NetBSD: Makefile,v 1.43 2007/08/03 22:40:00 mjl 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 asterisk to 1.2.23 * channels/chan_iax2.c: Don't create the Asterisk channel until we are starting the PBX on it. (ASA-2007-018) * channels/chan_agent.c: (closes issue #5866) Reported by: tyler Do not force channel format changes when a generator is present. The generator may have changed the formats itself and changing them back would cause issues. * channels/chan_sip.c: (closes issue #10236) Reported by: homesick Patches: rpid_1.4_75840.patch uploaded by homesick (license 91) Accept Remote Party ID on guest calls. * include/asterisk/app.h: We should not use C++ reserved words in API headers (closes issue #10266) * channels/chan_sip.c: Backport a fix for a memory leak that was fixed in trunk in reivision 76221 by rizzo. The memory used for the localaddr list was not freed during a configuration reload. * channels/chan_sip.c: (closes issue #10247) Reported by: fkasumovic Patches: chan_sip.patch uploaded by fkasumovic (license #101) Drop any peer realm authentication entries when reloading so multiple entries do not get added to the peer. * channels/chan_iax2.c: When processing full frames, take sequence number wraparound into account when deciding whether or not we need to request retransmissions by sending a VNAK. This code could cause VNAKs to be sent erroneously in some cases, and to not be sent in other cases when it should have been. (closes issue #10237, reported and patched by mihai) * channels/chan_iax2.c: When traversing the queue of frames for possible retransmission after receiving a VNAK, handle sequence number wraparound so that all frames that should be retransmitted actually do get retransmitted. (issue #10227, reported and patched by mihai) * apps/app_voicemail.c: Store prior to copy (closes issue #10193) * apps/app_queue.c: removed the word 'pissed' from ast_log(...)
2007-08-04 00:40:00 +02:00
DISTNAME= asterisk-1.2.23
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
Restore the 'old-releases' directory to MASTER_SITES so this package doesn't immediately break when the next version is released.
2005-09-02 19:23:06 +02:00
MASTER_SITES= http://ftp.digium.com/pub/asterisk/ \
Add a slash to the end of the old-releases URL. Fixes downloads for current package version 1.2.8, which is now old due to Asterisk release 1.2.9.
2006-06-11 01:56:18 +02:00
http://ftp.digium.com/pub/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
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
MAINTAINER= riz@NetBSD.org
HOMEPAGE= http://www.asterisk.org/
COMMENT= The Asterisk Software PBX
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
.include "../../mk/bsd.prefs.mk"
USE_TOOLS+= bison gmake perl:run
- 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
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
ASTMANDIR?= ${PREFIX}/man
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}|'
SUBST_SED.asterisk+= -e 's|^ *\(ASTCONFPATH\)=.*|\1 = ${ASTCONFPATH}|'
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
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
CONF_FILES+= ${ASTEXAMPLEDIR:Q}/${f:Q} ${ASTETCDIR:Q}/${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
- Fixed some pkglint warnings. - Replaced absolute directories like /usr/pkg and /var with ${PREFIX} and ${VARBASE}. - USE_TOOLS+=perl:run, since there is one Perl program installed with the package. - Bumped PKGREVISION.
2006-01-29 02:21:45 +01:00
MAKE_ENV+= VARBASE=${VARBASE:Q}
MAKE_FLAGS+= INSTALL_PREFIX=${PREFIX: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
# 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
Fixed the remaining pkglint -Wall warnings.
2006-01-29 02:30:20 +01:00
CONF_FILES+= ${ASTEXAMPLEDIR:Q}/${f:Q} ${ASTETCDIR:Q}/${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:
Fixed the remaining pkglint -Wall warnings.
2006-01-29 02:30:20 +01:00
${ECHO} "#include <sys/poll.h>" > ${WRKSRC:Q}/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"
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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