From 9d0252b7202164964c7332bd08869c05d096d098 Mon Sep 17 00:00:00 2001 From: russell Date: Mon, 21 Jul 2008 14:48:45 +0000 Subject: Merged revisions 132390 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132390 | russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.0@132391 f38db490-d61c-443f-a65b-d21fe96a405b --- build_tools/menuselect-deps.in | 1 + 1 file changed, 1 insertion(+) (limited to 'build_tools') diff --git a/build_tools/menuselect-deps.in b/build_tools/menuselect-deps.in index ce5b08d88..1cd18701e 100644 --- a/build_tools/menuselect-deps.in +++ b/build_tools/menuselect-deps.in @@ -30,6 +30,7 @@ PGSQL=@PBX_PGSQL@ POPT=@PBX_POPT@ PORTAUDIO=@PBX_PORTAUDIO@ PRI=@PBX_PRI@ +RESAMPLE=@PBX_RESAMPLE@ RADIUS=@PBX_RADIUS@ SPANDSP=@PBX_SPANDSP@ SPEEX=@PBX_SPEEX@ -- cgit v1.2.3