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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
113 lines
2.7 KiB
C
113 lines
2.7 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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*
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* \brief globally-accessible datastore information and callbacks
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*
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* \author Mark Michelson <mmichelson@digium.com>
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*/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/global_datastores.h"
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#include "asterisk/linkedlists.h"
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static void dialed_interface_destroy(void *data)
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{
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struct ast_dialed_interface *di = NULL;
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AST_LIST_HEAD(, ast_dialed_interface) *dialed_interface_list = data;
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if (!dialed_interface_list) {
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return;
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}
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AST_LIST_LOCK(dialed_interface_list);
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while ((di = AST_LIST_REMOVE_HEAD(dialed_interface_list, list)))
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ast_free(di);
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AST_LIST_UNLOCK(dialed_interface_list);
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AST_LIST_HEAD_DESTROY(dialed_interface_list);
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ast_free(dialed_interface_list);
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}
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static void *dialed_interface_duplicate(void *data)
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{
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struct ast_dialed_interface *di = NULL;
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AST_LIST_HEAD(, ast_dialed_interface) *old_list;
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AST_LIST_HEAD(, ast_dialed_interface) *new_list = NULL;
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if(!(old_list = data)) {
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return NULL;
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}
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if(!(new_list = ast_calloc(1, sizeof(*new_list)))) {
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return NULL;
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}
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AST_LIST_HEAD_INIT(new_list);
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AST_LIST_LOCK(old_list);
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AST_LIST_TRAVERSE(old_list, di, list) {
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struct ast_dialed_interface *di2 = ast_calloc(1, sizeof(*di2) + strlen(di->interface));
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if(!di2) {
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AST_LIST_UNLOCK(old_list);
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dialed_interface_destroy(new_list);
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return NULL;
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}
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strcpy(di2->interface, di->interface);
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AST_LIST_INSERT_TAIL(new_list, di2, list);
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}
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AST_LIST_UNLOCK(old_list);
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return new_list;
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}
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const struct ast_datastore_info dialed_interface_info = {
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.type = "dialed-interface",
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.destroy = dialed_interface_destroy,
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.duplicate = dialed_interface_duplicate,
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};
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static void secure_call_store_destroy(void *data)
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{
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struct ast_secure_call_store *store = data;
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ast_free(store);
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}
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static void *secure_call_store_duplicate(void *data)
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{
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struct ast_secure_call_store *old = data;
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struct ast_secure_call_store *new;
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if (!(new = ast_calloc(1, sizeof(*new)))) {
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return NULL;
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}
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new->signaling = old->signaling;
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new->media = old->media;
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return new;
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}
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const struct ast_datastore_info secure_call_info = {
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.type = "encrypt-call",
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.destroy = secure_call_store_destroy,
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.duplicate = secure_call_store_duplicate,
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};
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