From 857814f4354fb26255d4d5db6e06e90749e9bad0 Mon Sep 17 00:00:00 2001 From: Terry Wilson Date: Tue, 8 Jun 2010 05:29:08 +0000 Subject: 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 --- build_tools/menuselect-deps.in | 1 + 1 file changed, 1 insertion(+) (limited to 'build_tools/menuselect-deps.in') diff --git a/build_tools/menuselect-deps.in b/build_tools/menuselect-deps.in index ab22fbd64..2e8451b23 100644 --- a/build_tools/menuselect-deps.in +++ b/build_tools/menuselect-deps.in @@ -51,6 +51,7 @@ SPEEXDSP=@PBX_SPEEXDSP@ SPEEX_PREPROCESS=@PBX_SPEEX_PREPROCESS@ SQLITE3=@PBX_SQLITE3@ SQLITE=@PBX_SQLITE@ +SRTP=@PBX_SRTP@ SS7=@PBX_SS7@ OPENSSL=@PBX_OPENSSL@ SUPPSERV=@PBX_SUPPSERV@ -- cgit v1.2.3