mirror of
https://github.com/asterisk/asterisk.git
synced 2025-10-31 02:37:10 +00:00
527cf5a570
This removes references that are no longer needed due to automatic references created by module dependencies. In addition this removes most calls to ast_module_check as they were checking modules which are listed as dependencies. Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e
107 lines
2.8 KiB
C
107 lines
2.8 KiB
C
/*
|
|
* Asterisk -- An open source telephony toolkit.
|
|
*
|
|
* Copyright (C) 2015, 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.
|
|
*/
|
|
|
|
/*** MODULEINFO
|
|
<depend>pjproject</depend>
|
|
<depend>res_pjsip</depend>
|
|
<support_level>core</support_level>
|
|
***/
|
|
|
|
#include "asterisk.h"
|
|
|
|
#include <pjsip.h>
|
|
|
|
#include "asterisk/res_pjsip.h"
|
|
#include "asterisk/module.h"
|
|
|
|
/*!
|
|
* \brief Upgrade Contact URIs on outgoing SIP requests to SIPS if required.
|
|
*
|
|
* The rules being used here are according to RFC 3261 section 8.1.1.8. In
|
|
* brief, if the request URI is SIPS or the topmost Route header is SIPS,
|
|
* then the Contact header we send must also be SIPS.
|
|
*/
|
|
static pj_status_t sips_contact_on_tx_request(pjsip_tx_data *tdata)
|
|
{
|
|
pjsip_contact_hdr *contact;
|
|
pjsip_route_hdr *route;
|
|
pjsip_sip_uri *contact_uri;
|
|
|
|
contact = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CONTACT, NULL);
|
|
if (!contact) {
|
|
return PJ_SUCCESS;
|
|
}
|
|
|
|
contact_uri = pjsip_uri_get_uri(contact->uri);
|
|
if (PJSIP_URI_SCHEME_IS_SIPS(contact_uri)) {
|
|
/* If the Contact header is already SIPS, then we don't need to do anything */
|
|
return PJ_SUCCESS;
|
|
}
|
|
|
|
if (PJSIP_URI_SCHEME_IS_SIPS(tdata->msg->line.req.uri)) {
|
|
ast_debug(1, "Upgrading contact URI on outgoing SIP request to SIPS due to SIPS Request URI\n");
|
|
pjsip_sip_uri_set_secure(contact_uri, PJ_TRUE);
|
|
return PJ_SUCCESS;
|
|
}
|
|
|
|
route = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_ROUTE, NULL);
|
|
if (!route) {
|
|
return PJ_SUCCESS;
|
|
}
|
|
|
|
if (!PJSIP_URI_SCHEME_IS_SIPS(&route->name_addr)) {
|
|
return PJ_SUCCESS;
|
|
}
|
|
|
|
/* Our Contact header is not a SIPS URI, but our topmost Route header is. */
|
|
ast_debug(1, "Upgrading contact URI on outgoing SIP request to SIPS due to SIPS Route header\n");
|
|
pjsip_sip_uri_set_secure(contact_uri, PJ_TRUE);
|
|
|
|
return PJ_SUCCESS;
|
|
}
|
|
|
|
static pjsip_module sips_contact_module = {
|
|
.name = {"SIPS Contact", 12 },
|
|
.id = -1,
|
|
.priority = PJSIP_MOD_PRIORITY_TSX_LAYER - 2,
|
|
.on_tx_request = sips_contact_on_tx_request,
|
|
};
|
|
|
|
static int unload_module(void)
|
|
{
|
|
ast_sip_unregister_service(&sips_contact_module);
|
|
return 0;
|
|
}
|
|
|
|
static int load_module(void)
|
|
{
|
|
if (ast_sip_register_service(&sips_contact_module)) {
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
}
|
|
|
|
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "UAC SIPS Contact support",
|
|
.support_level = AST_MODULE_SUPPORT_CORE,
|
|
.load = load_module,
|
|
.unload = unload_module,
|
|
.load_pri = AST_MODPRI_APP_DEPEND,
|
|
.requires = "res_pjsip",
|
|
);
|