The large GULP->PJSIP renaming effort.

The general gist is to have a clear boundary between old SIP stuff and new SIP stuff by having the word "SIP" for old stuff and "PJSIP" for new stuff. Here's a brief rundown of the changes: * The word "Gulp" in dialstrings, functions, and CLI commands is now "PJSIP" * chan_gulp.c is now chan_pjsip.c * Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*" * All files that were "res_sip*" are now "res_pjsip*" * The "res_sip" directory is now "res_pjsip" * Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*" * The configuration file is now "pjsip.conf" instead of "res_sip.conf" * The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP" * CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as the starting word instead of "sip" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2025-09-02 19:16:15 +00:00 · 2013-07-30 18:14:50 +00:00
parent 895c8e0d2c
commit 735b30ad71
54 changed files with 369 additions and 369 deletions
--- a/res/res_pjsip_endpoint_identifier_anonymous.c
+++ b/res/res_pjsip_endpoint_identifier_anonymous.c
@@ -0,0 +1,125 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2013, Digium, Inc.
+ *
+ * Mark Michelson <mmichelson@digium.com>
+ * Joshua Colp <jcolp@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"
+
+static int get_endpoint_details(pjsip_rx_data *rdata, char *domain, size_t domain_size)
+{
+	pjsip_uri *from = rdata->msg_info.from->uri;
+	pjsip_sip_uri *sip_from;
+	if (!PJSIP_URI_SCHEME_IS_SIP(from) && !PJSIP_URI_SCHEME_IS_SIPS(from)) {
+		return -1;
+	}
+	sip_from = (pjsip_sip_uri *) pjsip_uri_get_uri(from);
+	ast_copy_pj_str(domain, &sip_from->host, domain_size);
+	return 0;
+}
+
+static int find_transport_in_use(void *obj, void *arg, int flags)
+{
+	struct ast_sip_transport *transport = obj;
+	pjsip_rx_data *rdata = arg;
+
+	if ((transport->state->transport == rdata->tp_info.transport) ||
+		(transport->state->factory && !pj_strcmp(&transport->state->factory->addr_name.host, &rdata->tp_info.transport->local_name.host) &&
+			transport->state->factory->addr_name.port == rdata->tp_info.transport->local_name.port)) {
+		return CMP_MATCH | CMP_STOP;
+	}
+
+	return 0;
+}
+
+static struct ast_sip_endpoint *anonymous_identify(pjsip_rx_data *rdata)
+{
+	char domain_name[64], id[AST_UUID_STR_LEN];
+	struct ast_sip_endpoint *endpoint;
+	RAII_VAR(struct ast_sip_domain_alias *, alias, NULL, ao2_cleanup);
+	RAII_VAR(struct ao2_container *, transports, NULL, ao2_cleanup);
+	RAII_VAR(struct ast_sip_transport *, transport, NULL, ao2_cleanup);
+
+	if (get_endpoint_details(rdata, domain_name, sizeof(domain_name))) {
+		return NULL;
+	}
+
+	/* Attempt to find the endpoint given the name and domain provided */
+	snprintf(id, sizeof(id), "anonymous@%s", domain_name);
+	if ((endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", id))) {
+		goto done;
+	}
+
+	/* See if an alias exists for the domain provided */
+	if ((alias = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "domain_alias", domain_name))) {
+		snprintf(id, sizeof(id), "anonymous@%s", alias->domain);
+		if ((endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", id))) {
+			goto done;
+		}
+	}
+
+	/* See if the transport this came in on has a provided domain */
+	if ((transports = ast_sorcery_retrieve_by_fields(ast_sip_get_sorcery(), "transport", AST_RETRIEVE_FLAG_MULTIPLE | AST_RETRIEVE_FLAG_ALL, NULL)) &&
+		(transport = ao2_callback(transports, 0, find_transport_in_use, rdata)) &&
+		!ast_strlen_zero(transport->domain)) {
+		snprintf(id, sizeof(id), "anonymous@%s", transport->domain);
+		if ((endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", id))) {
+			goto done;
+		}
+	}
+
+	/* Fall back to no domain */
+	endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", "anonymous");
+
+done:
+	if (endpoint) {
+		ast_debug(3, "Retrieved anonymous endpoint '%s'\n", ast_sorcery_object_get_id(endpoint));
+	}
+	return endpoint;
+}
+
+static struct ast_sip_endpoint_identifier anonymous_identifier = {
+	.identify_endpoint = anonymous_identify,
+};
+
+static int load_module(void)
+{
+	ast_sip_register_endpoint_identifier(&anonymous_identifier);
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	ast_sip_unregister_endpoint_identifier(&anonymous_identifier);
+	return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Anonymous endpoint identifier",
+		.load = load_module,
+		.unload = unload_module,
+		.load_pri = AST_MODPRI_DEFAULT,
+	       );