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authortwilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b>2010-06-08 05:29:08 +0000
committertwilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b>2010-06-08 05:29:08 +0000
commit9b1a36a294342fc418d9a359a4cf06bd90c4acb9 (patch)
treeecc27fc0db142ea1cd335a74cd1265f993fecd11 /build_tools
parent5f87b66641d86dbe7afec3b083016b2b1aceafc7 (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: http://svn.digium.com/svn/asterisk/trunk@268894 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'build_tools')
-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@