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asterisk/build_tools/menuselect-deps.in

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improve configure script to remember the previous value of each dependency in build_tools/menuselect-deps, so that (once it has been written) menuselect can use this information to warn the user when a previously met dependency is no longer met along the way, change tags used in configure script, menuselect-deps and code for various dependencies to be consistently named git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-04 15:07:54 +00:00
ALSA=@PBX_ALSA@
Move Asterisk-addons modules into the main Asterisk source tree. Someone asked yesterday, "is there a good reason why we can't just put these modules in Asterisk?". After a brief discussion, as long as the modules are clearly set aside in their own directory and not enabled by default, it is perfectly fine. For more information about why a module goes in addons, see README-addons.txt. chan_ooh323 does not currently compile as it is behind some trunk API updates. However, it will not build by default, so it should be okay for now. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30 16:40:38 +00:00
BLUETOOTH=@PBX_BLUETOOTH@
This patch adds a beanstalk CDR backend. Beanstalkd is a simple to use job queue. It provides a means to create multiple job queues called "tubes". Each tube can store multiple jobs, with varying priorities with the queue. Queue processing is available via a simple TCP socket or via well defined libraries, avaialble at https://github.com/kr/beanstalkd/wiki/client-libraries This module is based upon the beanstalk-client library, available for download at: https://github.com/deepfryed/beanstalk-client Change-Id: I5fe4089a34ab3b39230786d9bbfddafa56715f48
2017-10-16 17:46:02 -04:00
BEANSTALK=@PBX_BEANSTALK@
Replace res_ais with a new module, res_corosync. This patch removes res_ais and introduces a new module, res_corosync. The OpenAIS project is deprecated and is now just a wrapper around Corosync. This module provides the same functionality using the same core infrastructure, but without the use of the deprecated components. Technically res_ais could have been used with an AIS implementation other than OpenAIS, but that is the only one I know of that was ever used. Review: https://reviewboard.asterisk.org/r/1700/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-05 10:58:37 +00:00
COROSYNC=@PBX_COROSYNC@
Add configure script check for JACK. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-13 18:08:50 +00:00
CRYPTO=@PBX_CRYPTO@
Merged revisions 298960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298960 | tilghman | 2010-12-17 17:52:04 -0600 (Fri, 17 Dec 2010) | 20 lines Merged revisions 298957 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298957 | tilghman | 2010-12-17 17:30:55 -0600 (Fri, 17 Dec 2010) | 13 lines Merged revisions 298905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010) | 6 lines Let Asterisk find better backtrace information with libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will use 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. Review: https://reviewboard.asterisk.org/r/1055/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298961 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-18 00:08:13 +00:00
BFD=@PBX_BFD@
Merged revisions 242966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r242966 | tilghman | 2010-01-25 15:36:33 -0600 (Mon, 25 Jan 2010) | 2 lines Only rebuild parsers by an option in menuselect ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-25 21:38:33 +00:00
BISON=@PBX_BISON@
codecs: Add Codec 2 mode 2400. ASTERISK-26217 #close Change-Id: I1e45d8084683fab5f2b272bf35f4a149cea8b8d6
2016-07-19 20:14:21 +02:00
CODEC2=@PBX_CODEC2@
Thanks to the fine work of Russell Bryant and Dancho Lazarov, we now have autoconf and menuselect tools for Asterisk! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-04-24 17:11:45 +00:00
CURL=@PBX_CURL@
Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-12 17:27:55 +00:00
DAHDI=@PBX_DAHDI@
Merged revisions 298960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298960 | tilghman | 2010-12-17 17:52:04 -0600 (Fri, 17 Dec 2010) | 20 lines Merged revisions 298957 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298957 | tilghman | 2010-12-17 17:30:55 -0600 (Fri, 17 Dec 2010) | 13 lines Merged revisions 298905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010) | 6 lines Let Asterisk find better backtrace information with libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will use 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. Review: https://reviewboard.asterisk.org/r/1055/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298961 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-18 00:08:13 +00:00
DLADDR=@PBX_DLADDR@
Merged revisions 242966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r242966 | tilghman | 2010-01-25 15:36:33 -0600 (Mon, 25 Jan 2010) | 2 lines Only rebuild parsers by an option in menuselect ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-25 21:38:33 +00:00
FLEX=@PBX_FLEX@
massive rework of configure script, and split of AST_EXT_LIB into separate macros, reducing code duplication git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-05 19:59:34 +00:00
FREETDS=@PBX_FREETDS@
remove the PBX_ODBC logic from the configure script, and add GENERIC_ODCB logic that includes copying the relevant LIB and INCLUDE data from either UnixODBC or iODBC, based on which was found; if both were found, prefer UnixODBC this stops modules from being linked against both sets of libraries on systems that have both installed git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-15 20:18:53 +00:00
GENERIC_ODBC=@PBX_GENERIC_ODBC@
Replace minimime with superior GMime library so that the entire contents of an http post are not read into memory. This does introduce a dependency on the GMime library for handling HTTP POSTs, but it is available in most distros. If the library is present, then the compile flag for ENABLE_UPLOADS is enabled by default in menuselect. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-17 22:10:06 +00:00
GMIME=@PBX_GMIME@
Add configure script check for JACK. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-13 18:08:50 +00:00
GNU_LD=@GNU_LD@
sort entries in this file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-07 05:29:31 +00:00
GSM=@PBX_GSM@
Enable usage of system-provided iLBC library. The WebRTC version of the iLBC codec is now package as a library and is available on some platforms. This patch allows codec_ilbc to be built against that library if it is present. Review: https://reviewboard.asterisk.org/r/1964/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-23 21:27:56 +00:00
ILBC=@PBX_ILBC@
Merged revisions 58947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58947 | russell | 2007-03-15 18:53:26 -0500 (Thu, 15 Mar 2007) | 3 lines Add configure script checking for GTK2 and some additional Makefile targets to support gmenuselect ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-15 23:56:10 +00:00
GTK2=@PBX_GTK2@
Merge in changes that allow Asterisk to be built against the Hoard memory allocator. See doc/hoard.txt for more details. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-03 16:14:14 +00:00
HOARD=@PBX_HOARD@
Add Calendaring support for Asterisk This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS Exchange calendars. Exchange support has only been tested on Exchange Server 2k3 and does not support forms-based authentication at this time (patches *very* welcome). Exchange support is also currently missing the ability to return a list of a meting's attendees (again, patches are very, very welcome). Features include: Querying a calendar for events over a specific time range Checking a calendar's busy status via the dialplan Writing calendar events via the dialplan (CalDAV and Exchange only) Handling calendar event notifications through the dialplan (closes issue #14771) Tested by: lmadsen, twilson, Shivaprakash Review: https://reviewboard.asterisk.org/r/58 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 19:57:18 +00:00
ICAL=@PBX_ICAL@
implement "configure" checks for libiconv, and add the iconv dependency for func_iconv. This fixes some build issues on CYGWIN and FreeBSD and probably other platforms where libiconv is not there by default git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-01 22:08:32 +00:00
ICONV=@PBX_ICONV@
sort entries in this file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-07 05:29:31 +00:00
IKSEMEL=@PBX_IKSEMEL@
IMAP_TK=@PBX_IMAP_TK@
All ODBC parts can now use either unixodbc or iodbc. This allows for the ODBC parts to work on OpenBSD as well. 99.99% of the work is done by seanbright (bow, bow) and I actually did nothing but test and yell at him that it still didn't work :) Thanks for helping out ! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146925 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-06 23:14:33 +00:00
IODBC=@PBX_IODBC@
Add configure script check for JACK. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-13 18:08:50 +00:00
JACK=@PBX_JACK@
Add JSON API for Asterisk. This provides a JSON API by pulling in and wrapping the Jansson JSON library[1]. The Asterisk API basically mirrors the Jansson functionality, with a few minor tweaks. * Some names have been asteriskified to protect the innocent. * Jansson provides both reference-stealing and reference-borrowing versions of several API's. The Asterisk API is exclusively reference-stealing for operations that put elements into arrays and objects. * No support for doubles, since we usually don't need that. * Coming along for the ride is the ast_test_validate macro, which made the unit tests much easier to write. [1]: http://www.digip.org/jansson/ (issue ASTERISK-20887) (closes issue ASTERISK-20888) Review: https://reviewboard.asterisk.org/r/2264/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-11 22:31:42 +00:00
JANSSON=@PBX_JANSSON@
Add the bucket API. Bucket is a URI based API for the creation, retrieval, updating, and deletion of "buckets" and files contained within them. Review: https://reviewboard.asterisk.org/r/2715/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23 21:49:47 +00:00
URIPARSER=@PBX_URIPARSER@
Add kqueue(2) implementation to Asterisk in various places. This will save a considerable amount of CPU on the BSDs, including Mac OS X, as it eliminates several places in the code that we previously used a busy loop. Additionally, this adds a res_timing interface, using kqueue timers. Review: https://reviewboard.asterisk.org/r/543/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-13 05:37:31 +00:00
KQUEUE=@PBX_KQUEUE@
Add res_config_ldap for realtime LDAP engine. (closes issue #5768) Reported by: mguesdon Patches: res_config_ldap-v0.7.tar.gz uploaded by mguesdon (license 121) res_ldap.conf.sample uploaded by suretec (license 70) asterisk-v3.1.4.ldif uploaded by suretec (license 70) asterisk-v3.1.4.schema uploaded by suretec (license 70) Tested by: oej, mguesdon, suretec, cthorner git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-22 22:33:20 +00:00
LDAP=@PBX_LDAP@
Enable usage of system-provided NetBSD editline library if available. This patch changes the Asterisk configure script and build system to detect the presence of the NetBSD editline library (libedit) on the system. If it is found, it will be used in preference to the version included in the Asterisk source tree. (closes issue ASTERISK-18725) Reported by: Jeffrey C. Ollie Review: https://reviewboard.asterisk.org/r/1528/ Patches: 0001-Allow-linking-building-against-an-external-editline.patch uploaded by jcollie (license #5373) (heavily modified by kpfleming) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-25 12:21:54 +00:00
LIBEDIT=@PBX_LIBEDIT@
Add libjwt to third-party The current STIR/SHAKEN implementation is not currently usable due to encryption issues. Rather than trying to futz with OpenSSL and the the current code, we can take advantage of the existing capabilities of libjwt but we first need to add it to the third-party infrastructure already in place for jansson and pjproject. A few tweaks were also made to the third-party infrastructure as a whole. The jansson "dest" install directory was renamed "dist" to better match convention, and the third-party Makefile was updated to clean all product directories not just the ones currently in use. Resolves: #349
2023-09-21 11:12:31 -06:00
LIBJWT=@PBX_LIBJWT@
Add Calendaring support for Asterisk This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS Exchange calendars. Exchange support has only been tested on Exchange Server 2k3 and does not support forms-based authentication at this time (patches *very* welcome). Exchange support is also currently missing the ability to return a list of a meting's attendees (again, patches are very, very welcome). Features include: Querying a calendar for events over a specific time range Checking a calendar's busy status via the dialplan Writing calendar events via the dialplan (CalDAV and Exchange only) Handling calendar event notifications through the dialplan (closes issue #14771) Tested by: lmadsen, twilson, Shivaprakash Review: https://reviewboard.asterisk.org/r/58 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 19:57:18 +00:00
LIBXML2=@PBX_LIBXML2@
xml.c, config,c: Add stylesheets and variable list string parsing Added functions to open, close, and apply XML Stylesheets to XML documents. Although the presence of libxslt was already being checked by configure, it was only happening if xmldoc was enabled. Now it's checked regardless. Added ability to parse a string consisting of comma separated name/value pairs into an ast_variable list. The reverse of ast_variable_list_join(). Change-Id: I1e1d149be22165a1fb8e88e2903a36bba1a6cf2e
2022-03-02 07:57:26 -07:00
LIBXSLT=@PBX_LIBXSLT@
build: Add download capability for external packages The DPMA and g729a, silk, siren7 and siren14 codecs hosted at http://downloads.digium.com/pub/telephony/ are now listed in the "External" sections of the "Resource Modules" and "Codec Translators" pages in menuselect. Any that are selected will automatically be downloaded and installed when "make install" is run. Their LICENSE and README (if avaialble) files will be installed to ASTVARLIBDIR/documentation/thirdparty/<product_name>. Example use with codecs: The codecs/codecs.xml file is a menuselect style xml file that lists the codecs to be included. Their support levels are 'external', which triggers the download and install, and defaultenabled is no. Also because codec_g729a is actually in a directory named codec_g729 on the download server, the newly added 'member_data' element is used to override the default of the directory name being the package name. You can use the 'directory_name' attribute to keep default base URL (http://downloads.digium.com/pub/telephony/) but use the new directory, or you use the 'remote_url' attribute to specify a full URL to the download directory. In this case, you must still follow the same subdirectory naming conventions as that used for the packages located at 'http://downloads.digium.com/pub/telephony'. A new configure option '--with-externals-cache' was added and like '--with-sounds-cache' it allows the installer to cache tarballs so they're not downloaded every time. To assist with the download and install process, each external package now has a manifest.xml file that, among other things, contains a package version and checksums for each file in the tarball. The manifest is saved to both the cache directory and ASTMODDIR and together with the manifest.xml on the downloads site, tells the install scripts whether a download and/or update is needed. bash and xmlstarlet are required for downloader operation. If they're not installed, the external items in menuselect will be unavailable. Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a
2016-08-01 19:55:33 -06:00
XMLSTARLET=@PBX_XMLSTARLET@
BASH=@PBX_BASH@
Add pbx_lua as a method of doing extensions Reported by: mnicholson Patch by: mnicholson Closes issue #11140 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-02 15:36:34 +00:00
LUA=@PBX_LUA@
Move Asterisk-addons modules into the main Asterisk source tree. Someone asked yesterday, "is there a good reason why we can't just put these modules in Asterisk?". After a brief discussion, as long as the modules are clearly set aside in their own directory and not enabled by default, it is perfectly fine. For more information about why a module goes in addons, see README-addons.txt. chan_ooh323 does not currently compile as it is behind some trunk API updates. However, it will not build by default, so it should be okay for now. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-30 16:40:38 +00:00
MYSQLCLIENT=@PBX_MYSQLCLIENT@
sort entries in this file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-07 05:29:31 +00:00
NETSNMP=@PBX_NETSNMP@
NEWT=@PBX_NEWT@
Add Calendaring support for Asterisk This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS Exchange calendars. Exchange support has only been tested on Exchange Server 2k3 and does not support forms-based authentication at this time (patches *very* welcome). Exchange support is also currently missing the ability to return a list of a meting's attendees (again, patches are very, very welcome). Features include: Querying a calendar for events over a specific time range Checking a calendar's busy status via the dialplan Writing calendar events via the dialplan (CalDAV and Exchange only) Handling calendar event notifications through the dialplan (closes issue #14771) Tested by: lmadsen, twilson, Shivaprakash Review: https://reviewboard.asterisk.org/r/58 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28 19:57:18 +00:00
NEON=@PBX_NEON@
Ensure that libneon > 0.29.0 is installed for res_calendar_ews This uses a modified version of pabelanger's patch that checks for NTLM support instead, which was added in 0.29.0 which is what is required for res_calendar_ews. (closes issue #17391) Reported by: loloski Patches: issue17391.patch.v2 uploaded by pabelanger (license 224) Tested by: twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-26 05:33:11 +00:00
NEON29=@PBX_NEON29@
massive rework of configure script, and split of AST_EXT_LIB into separate macros, reducing code duplication git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-05 19:59:34 +00:00
OGG=@PBX_OGG@
configure: Add autoconf check for libopus. Because opus transcoding support cannot be included in the standard Asterisk distribution, a few codec_opus implementations have popped up. To make it easier for people to drop in opus support in their own installations, this patch adds configure checks for libopus. Review: https://reviewboard.asterisk.org/r/4106/ ........ Merged revisions 426234 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-27 17:55:43 +00:00
OPUS=@PBX_OPUS@
configure: Add autoconf check for libopusfile. This check is being added to make it easier for end-users of third party open source Opus modules. This was removed by ASTERISK-26426 but only the module needed to be removed. Change-Id: I62b9cd0c4fa8a77596ab0e042948a643a1152677
2017-11-06 04:21:12 -05:00
OPUSFILE=@PBX_OPUSFILE@
sort entries in this file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-07 05:29:31 +00:00
OSPTK=@PBX_OSPTK@
massive rework of configure script, and split of AST_EXT_LIB into separate macros, reducing code duplication git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-05 19:59:34 +00:00
PGSQL=@PBX_PGSQL@
Switch to using external pjproject libraries. ICE/STUN/TURN support in res_rtp_asterisk is also now optional. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-03-12 19:08:59 +00:00
PJPROJECT=@PBX_PJPROJECT@
sort entries in this file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-07 05:29:31 +00:00
POPT=@PBX_POPT@
Add configure script checking for portaudio. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95383 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-31 15:20:30 +00:00
PORTAUDIO=@PBX_PORTAUDIO@
sort entries in this file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-07 05:29:31 +00:00
PRI=@PBX_PRI@
added openr2 to menuselect-deps.in, recent commit in menuselect made me realize this was never done but was working anyways also added support for skip category request feature of openr2 and updated chan_dahdi.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-14 06:13:48 +00:00
OPENR2=@PBX_OPENR2@
Remove libresample from the Asterisk source tree. It is now available in its own repository, and must be installed like any other library for Asterisk to use. The two modules that require it are codec_resample and app_jack. To install libresample: $ svn co http://svn.digium.com/svn/libresample/trunk libresample $ cd libresample $ ./configure $ make $ sudo make install This code is currently in our own repository because the build system did not include the appropriate targets for building a dynamic library or for installing the library. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-21 14:47:41 +00:00
RESAMPLE=@PBX_RESAMPLE@
Binaural synthesis (confbridge): Adds libfftw3 as dependency. Adds libfftw3 to the build chain that is is going to be used for binaural synthesis by bridge_softmix. ASTERISK-26292 Change-Id: Iedc2f174e4ccb39ae5d9e698e339c6a17155867b
2016-08-12 18:22:40 +02:00
FFTW3=@PBX_FFTW3@
massive rework of configure script, and split of AST_EXT_LIB into separate macros, reducing code duplication git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-05 19:59:34 +00:00
RADIUS=@PBX_RADIUS@
Conflict kqueue on OS X, since it doesn't work there yet, anyway. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271657 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-21 22:41:00 +00:00
LAUNCHD=@PBX_LAUNCHD@
Binaural synthesis (confbridge): Adds utils/conf_bridge_binaural_hrir_importer Adds the import tool for converting a HRIR database to hrirs.h ASTERISK-26292 Change-Id: I51eb31b54c23ffd9b544bdc6a09d20c112c8a547
2016-12-22 16:42:46 +01:00
SNDFILE=@PBX_SNDFILE@
Add a configure script check for spandsp git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-02 15:43:40 +00:00
SPANDSP=@PBX_SPANDSP@
sort entries in this file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-07 05:29:31 +00:00
SPEEX=@PBX_SPEEX@
Merged revisions 98951 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98951 | file | 2008-01-15 21:13:27 -0400 (Tue, 15 Jan 2008) | 4 lines Add autoconf logic for speexdsp. Later versions use a separate library for some things so we need to use it if present in codec_speex. (closes issue #11693) Reported by: yzg ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-16 01:17:25 +00:00
SPEEXDSP=@PBX_SPEEXDSP@
Merged revisions 115327 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115327 | file | 2008-05-05 19:10:05 -0300 (Mon, 05 May 2008) | 2 lines Make sure that either the main speex library contains preprocess functions or that speexdsp does. If both fail then speex stuff can not be built. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-05 22:13:57 +00:00
SPEEX_PREPROCESS=@PBX_SPEEX_PREPROCESS@
Merge changes from team/russell/sqlite: * Add new module, cdr_sqlite3_custom which allows logging custom CDRs into a SQLite3 database. (issue #7149, alerios) * Add new module, res_config_sqlite, which adds realtime database configuration support for SQLite version 2. I decided that this was ok since we didn't have any realtime support for version 3. If someone ports this to version 3, then version 2 support can be removed or marked deprecated. (issue #7790, rbarun_proformatique) * Mark cdr_sqlite as deprecated in favor of cdr_sqlite3_custom. Also, note that there were other modules on the bug tracker that did not make the cut because they provided some duplicated functionality. Those are: * cdr_sqlite3 (issue #6754, moy) * cdr_sqlite3 (issue #8694, bsd) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-13 21:22:33 +00:00
SQLITE3=@PBX_SQLITE3@
Add SRTP support for Asterisk After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-08 05:29:08 +00:00
SRTP=@PBX_SRTP@
Add configure script check for JACK. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-13 18:08:50 +00:00
SS7=@PBX_SS7@
improve configure script to remember the previous value of each dependency in build_tools/menuselect-deps, so that (once it has been written) menuselect can use this information to warn the user when a previously met dependency is no longer met along the way, change tags used in configure script, menuselect-deps and code for various dependencies to be consistently named git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-04 15:07:54 +00:00
OPENSSL=@PBX_OPENSSL@
Add a new module, cdr_syslog, which allows writing CDRs to syslog. The original patch for this was written by Brett Bryant, and I split it out into it's own module. (closes issue #12876) Reported by: bbryant Patches: 06162008_cdr_custom_syslog.diff uploaded by bbryant (license 36) 05212009_cdr_syslog.patch uploaded by seanbright (license 71) Tested by: seanbright Review: https://reviewboard.asterisk.org/r/297/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26 22:08:05 +00:00
SYSLOG=@PBX_SYSLOG@
sort entries in this file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-07 05:29:31 +00:00
TONEZONE=@PBX_TONEZONE@
dns: Add core DNS API + unit tests and res_resolver_unbound module + unit tests. This change adds an abstracted core DNS API which resembles the API described here[1]. The API provides a pluggable mechanism for resolvers and also a consistent view for records. Both synchronous and asynchronous queries are supported. This change also adds a res_resolver_unbound module which uses the libunbound library to provide resolution. Unit tests have also been written for all of the above to confirm the API and functionality. ASTERISK-24834 #close Reported by: Matt Jordan ASTERISK-24836 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4474/ Review: https://reviewboard.asterisk.org/r/4512/ [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+DNS+API git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-25 12:32:26 +00:00
UNBOUND=@PBX_UNBOUND@
massive rework of configure script, and split of AST_EXT_LIB into separate macros, reducing code duplication git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-05 19:59:34 +00:00
UNIXODBC=@PBX_UNIXODBC@
sort entries in this file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-07 05:29:31 +00:00
VORBIS=@PBX_VORBIS@
improve configure script to remember the previous value of each dependency in build_tools/menuselect-deps, so that (once it has been written) menuselect can use this information to warn the user when a previously met dependency is no longer met along the way, change tags used in configure script, menuselect-deps and code for various dependencies to be consistently named git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-04 15:07:54 +00:00
WINARCH=@PBX_WINARCH@
sort entries in this file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-07-07 05:29:31 +00:00
ZLIB=@PBX_ZLIB@
Merge the changes from the res_timing_timerfd branch. This provides a new timing interface. In order to use it, you must be running a Linux with a kernel version of 2.6.25 or newer and glibc 2.8 or newer. This timing interface is a good alternative if a timing source is necessary (e.g. for IAX trunking) but DAHDI is otherwise unnecessary for the system. For now, this commit contains the actual work done in the res_timing_timerfd branch. There are no notices in the README or CHANGES files yet, but they will be added in my next commit. The timing API of Asterisk also needs to have a bit of work done with regards to choosing which timing interface to use. This commit makes the choice a build-time decision, by only allowing one of the timer interfaces to be chosen in menuselect. It would be preferable if the choice could be made at run-time, however. The preferred timing interface could be loaded and tested, and if it does not work, choice number two may be used instead. That sort of thing. That is beyond the scope of work in this branch though. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-19 19:37:32 +00:00
TIMERFD=@PBX_TIMERFD@
Simplify build system architecture optimization This change to the build system rips out any usage of PROC along with architecture-specific optimizations in favor of using -march=native where it is supported. This fixes broken builds on 64bit Intel systems and results in better optimized code on systems running GCC 4.2+. Review: https://reviewboard.asterisk.org/r/1852/ (closes issue ASTERISK-19462) ........ Merged revisions 361955 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361956 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-12 15:25:47 +00:00
NATIVE_ARCH=@AST_NATIVE_ARCH@
Build: Added testing compiler to support the system sanitizes In older versions of the compiler was not sanitizes. Compilers other than GCC can not support the Usan and TSAN or have other options for *FLAGS. ASTERISK-25767 #close Reported by: Badalyan Vyacheslav Tested by: Badalyan Vyacheslav Change-Id: Iefce6608221fa87884b82ae3cb5649b7b1804916
2016-02-10 05:40:32 +00:00
HAVE_ADDRESS_SANITIZER=@AST_ADDRESS_SANITIZER@
HAVE_LEAK_SANITIZER=@AST_LEAK_SANITIZER@
HAVE_THREAD_SANITIZER=@AST_THREAD_SANITIZER@
HAVE_UNDEFINED_SANITIZER=@AST_UNDEFINED_SANITIZER@
Create --disable-binary-modules option. This new option can be passed for ./configure or ./tests/CI/buildAsterisk.sh to prevent download/install of binary modules. Normally enabling the categories MENUSELECT_CODECS or MENUSELECT_RES will result in binary modules being enabled even if the build target is incompatible with those modules. This includes CI scripts which enable categories before disabling specific modules. If more binary modules are offered in the future this will help avoid accidentally downloading them if unwanted or incompatible. Adding a binary module will only require creating a new menuselect entry similar to the existing ones, it will not be necessary to modify the CI scripts. Change-Id: I6b1bd1c75a2e48f05b8b8a45b7a7a2d00a079166
2018-08-26 14:18:42 -04:00
NO_BINARY_MODULES=@PBX_NO_BINARY_MODULES@
Add C++ Standard detection to configure and fix a new C++20 compile issue * The autoconf-archive package contains macros useful for detecting C++ standard and testing other C++ capabilities but that package was never included in the install_prereq script so many existing build environments won't have it. Even if it is installed, older versions won't newer C++ standards and will actually cause an error if you try to test for that version. To make it available for those environments, the ax_cxx_compile_stdcxx.m4 macro has copied from the latest release of autoconf-archive into the autoconf directory. * A convenience wrapper(ast_cxx_check_std) around ax_cxx_compile_stdcxx was also added so checking the standard version and setting the asterisk-specific PBX_ variables becomes a one-liner: `AST_CXX_CHECK_STD([std], [force_latest_std])`. Calling that with a version of `17` for instance, will set PBX_CXX17 to 0 or 1 depending on whether the current c++ compiler supports stdc++17. HAVE_CXX17 will also be 'defined" or not depending on the result. * C++ compilers hardly ever default to the latest standard they support. g++ version 14 for instance supports up to C++23 but only uses C++17 by default. If you want to use C++23, you have to add `-std=gnu++=23` to the g++ command line. If you set the second argument of AST_CXX_CHECK_STD to "yes", the macro will automatically keep the highest `-std=gnu++` value that worked and pass that to the Makefiles. * The autoconf-archive package was added to install_prereq for future use. * Updated configure.ac to use AST_CXX_CHECK_STD() to check for C++ versions 11, 14, 17, 20 and 23. * Updated configure.ac to accept the `--enable-latest-cxx-std` option which will set the second option to AST_CXX_CHECK_STD() to "yes". The default is "no". * ast_copy_string() in strings.h declares the 'sz' variable as volatile and does an `sz--` on it later. C++20 no longer allows the `++` and `--` increment and decrement operators to be used on variables declared as volatile however so that was changed to `sz -= 1`.
2025-01-03 15:38:52 -07:00
CXX11=@PBX_CXX11@
CXX14=@PBX_CXX14@
CXX17=@PBX_CXX17@
CXX20=@PBX_CXX20@
CXX23=@PBX_CXX23@
res_srtp: Add menuselect options to enable AES_192, AES_256 and AES_GCM UserNote: Options are now available in the menuselect "Resource Modules" category that allow you to enable the AES_192, AES_256 and AES_GCM cipher suites in res_srtp. Of course, libsrtp and OpenSSL must support them but modern versions do. Previously, the only way to enable them was to set the CFLAGS environment variable when running ./configure. The default setting is to disable them preserving existing behavior.
2025-08-05 08:10:29 -06:00
HAVE_SRTP_192=@PBX_SRTP_192@
HAVE_SRTP_256=@PBX_SRTP_256@
HAVE_SRTP_GCM=@PBX_SRTP_GCM@
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