asterisk/include/asterisk/global_datastores.h
Terry Wilson 857814f435 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
2010-06-08 05:29:08 +00:00

42 lines
1.1 KiB
C

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