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authorTerry Wilson <twilson@digium.com>2010-06-08 05:29:08 +0000
committerTerry Wilson <twilson@digium.com>2010-06-08 05:29:08 +0000
commit857814f4354fb26255d4d5db6e06e90749e9bad0 (patch)
treeecc27fc0db142ea1cd335a74cd1265f993fecd11 /build_tools/menuselect-deps.in
parentebbf166c2d15fd233ee307e760b2a88c46d19f6b (diff)
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
Diffstat (limited to 'build_tools/menuselect-deps.in')
-rw-r--r--build_tools/menuselect-deps.in1
1 files changed, 1 insertions, 0 deletions
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@