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857814f435
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
42 lines
1.1 KiB
C
42 lines
1.1 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2007, Digium, Inc.
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*
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* Mark Michelson <mmichelson@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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* \brief globally accessible channel datastores
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* \author Mark Michelson <mmichelson@digium.com>
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*/
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#ifndef _ASTERISK_GLOBAL_DATASTORE_H
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#define _ASTERISK_GLOBAL_DATASTORE_H
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#include "asterisk/channel.h"
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extern const struct ast_datastore_info dialed_interface_info;
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extern const struct ast_datastore_info secure_call_info;
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struct ast_dialed_interface {
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AST_LIST_ENTRY(ast_dialed_interface) list;
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char interface[1];
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};
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struct ast_secure_call_store {
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unsigned int signaling:1;
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unsigned int media:1;
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};
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#endif
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