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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
52 lines
968 B
Text
52 lines
968 B
Text
ASOUND=@PBX_ALSA@
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CRYPTO=@PBX_CRYPTO@
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CURL=@PBX_CURL@
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DAHDI=@PBX_DAHDI@
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FREETDS=@PBX_FREETDS@
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GMIME=@PBX_GMIME@
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GNU_LD=@GNU_LD@
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GSM=@PBX_GSM@
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GTK2=@PBX_GTK2@
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GTK=@PBX_GTK@
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H323=@PBX_H323@
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ICONV=@PBX_ICONV@
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IKSEMEL=@PBX_IKSEMEL@
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IMAP_TK=@PBX_IMAP_TK@
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ISDNNET=@PBX_ISDNNET@
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IXJUSER=@PBX_IXJUSER@
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JACK=@PBX_JACK@
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LDAP=@PBX_LDAP@
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LTDL=@PBX_LTDL@
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LUA=@PBX_LUA@
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MISDN=@PBX_MISDN@
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NBS=@PBX_NBS@
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NETSNMP=@PBX_NETSNMP@
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NEWT=@PBX_NEWT@
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OGG=@PBX_OGG@
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OPENH323=@PBX_OPENH323@
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OSPTK=@PBX_OSPTK@
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OSSAUDIO=@PBX_OSS@
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PGSQL=@PBX_PGSQL@
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POPT=@PBX_POPT@
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PORTAUDIO=@PBX_PORTAUDIO@
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PRI=@PBX_PRI@
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RESAMPLE=@PBX_RESAMPLE@
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AIS=@PBX_AIS@
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RADIUS=@PBX_RADIUS@
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SPANDSP=@PBX_SPANDSP@
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SPEEX=@PBX_SPEEX@
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SPEEXDSP=@PBX_SPEEXDSP@
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SPEEX_PREPROCESS=@PBX_SPEEX_PREPROCESS@
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SQLITE3=@PBX_SQLITE3@
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SQLITE=@PBX_SQLITE@
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SS7=@PBX_SS7@
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SSL=@PBX_OPENSSL@
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SUPPSERV=@PBX_SUPPSERV@
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TONEZONE=@PBX_TONEZONE@
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UNIXODBC=@PBX_UNIXODBC@
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USB=@PBX_USB@
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VORBIS=@PBX_VORBIS@
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VPBAPI=@PBX_VPB@
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WINARCH=@WINARCH@
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ZLIB=@PBX_ZLIB@
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