Importing release summary for 11.0.0 release.

git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.0.0@375354 65c4cc65-6c06-0410-ace0-fbb531ad65f3

.lastclean

@@ -0,0 +1 @@
+40

.version

@@ -0,0 +1 @@
+11.0.0

channels/chan_iax2.c

@@ -1092,7 +1092,6 @@ static void iax_pvt_callid_new(int callno)
 	struct ast_callid *callid = ast_create_callid();
 	char buffer[AST_CALLID_BUFFER_LENGTH];
 	ast_callid_strnprint(buffer, sizeof(buffer), callid);
-	ast_log(LOG_NOTICE, "iax_pvt_callid_new created and set %s\n", buffer);
 	iax_pvt_callid_set(callno, callid);
 	ast_callid_unref(callid);
 }

channels/chan_motif.c

@@ -1729,6 +1729,7 @@ static struct ast_channel *jingle_request(const char *type, struct ast_format_ca
 	struct jingle_session *session;
 	struct ast_channel *chan;
 	enum jingle_transport transport = JINGLE_TRANSPORT_NONE;
+	struct ast_rtp_engine_ice *ice;
 	AST_DECLARE_APP_ARGS(args,
 			     AST_APP_ARG(name);
 			     AST_APP_ARG(target);
@@ -1832,6 +1833,15 @@ static struct ast_channel *jingle_request(const char *type, struct ast_format_ca
 		jingle_enable_video(session);
 	}
 
+	/* As this is outgoing set ourselves as controlling */
+	if (session->rtp && (ice = ast_rtp_instance_get_ice(session->rtp))) {
+		ice->ice_lite(session->rtp);
+	}
+
+	if (session->vrtp && (ice = ast_rtp_instance_get_ice(session->vrtp))) {
+		ice->ice_lite(session->vrtp);
+	}
+
 	/* We purposely don't decrement the session here as there is a reference on the channel */
 	ao2_link(endpoint->state->sessions, session);
 
@@ -1939,14 +1949,10 @@ static int jingle_interpret_ice_udp_transport(struct jingle_session *session, ik
 		return -1;
 	}
 
-	if (ast_strlen_zero(ufrag) || ast_strlen_zero(pwd)) {
-		jingle_queue_hangup_with_cause(session, AST_CAUSE_PROTOCOL_ERROR);
-		ast_log(LOG_ERROR, "Invalid ICE-UDP transport information received on session '%s', ufrag or pwd not present\n", session->sid);
-		return -1;
+	if (!ast_strlen_zero(ufrag) && !ast_strlen_zero(pwd)) {
+		ice->set_authentication(rtp, ufrag, pwd);
 	}
 
-	ice->set_authentication(rtp, ufrag, pwd);
-
 	for (candidate = iks_child(transport); candidate; candidate = iks_next(candidate)) {
 		char *component = iks_find_attrib(candidate, "component"), *foundation = iks_find_attrib(candidate, "foundation");
 		char *generation = iks_find_attrib(candidate, "generation"), *id = iks_find_attrib(candidate, "id");
@@ -2102,7 +2108,8 @@ static int jingle_interpret_content(struct jingle_session *session, ikspak *pak)
 		struct ast_rtp_instance *rtp = NULL;
 		iks *description, *transport;
 
-		if (strcmp(iks_name(content), "content")) {
+		/* Ignore specific parts if they are known not to be useful */
+		if (!strcmp(iks_name(content), "conference-info")) {
 			continue;
 		}
 

channels/chan_sip.c

@@ -7287,7 +7287,8 @@ static int initialize_udptl(struct sip_pvt *p)
 	return 0;
 }
 
-int ast_sipinfo_send(
+
+static int sipinfo_send(
 		struct ast_channel *chan,
 		struct ast_variable *headers,
 		const char *content_type,
@@ -33162,6 +33163,12 @@ static const struct ast_data_entry sip_data_providers[] = {
 	AST_DATA_ENTRY("asterisk/channel/sip/peers", &peers_data_provider),
 };
 
+static const struct ast_sip_api_tech chan_sip_api_provider = {
+	.version = AST_SIP_API_VERSION,
+	.name = "chan_sip",
+	.sipinfo_send = sipinfo_send,
+};
+
 /*! \brief PBX load module - initialization */
 static int load_module(void)
 {
@@ -33171,6 +33178,10 @@ static int load_module(void)
 		return AST_MODULE_LOAD_FAILURE;
 	}
 
+	if (ast_sip_api_provider_register(&chan_sip_api_provider)) {
+		return AST_MODULE_LOAD_FAILURE;
+	}
+
 	/* the fact that ao2_containers can't resize automatically is a major worry! */
 	/* if the number of objects gets above MAX_XXX_BUCKETS, things will slow down */
 	peers = ao2_t_container_alloc(HASH_PEER_SIZE, peer_hash_cb, peer_cmp_cb, "allocate peers");
@@ -33332,6 +33343,8 @@ static int unload_module(void)
 	struct ao2_iterator i;
 	int wait_count;
 
+	ast_sip_api_provider_unregister();
+
 	ast_websocket_remove_protocol("sip", sip_websocket_callback);
 
 	network_change_event_unsubscribe();
@@ -33507,7 +33520,7 @@ static int unload_module(void)
 	return 0;
 }
 
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Session Initiation Protocol (SIP)",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Session Initiation Protocol (SIP)",
 		.load = load_module,
 		.unload = unload_module,
 		.reload = reload,

channels/chan_sip.exports.in

@@ -1,6 +0,0 @@
-{
-	global:
-		LINKER_SYMBOL_PREFIX*ast_sipinfo_send;
-	local:
-		*;
-};

include/asterisk/sip_api.h

@@ -26,6 +26,16 @@ extern "C" {
 #include "asterisk/optional_api.h"
 #include "asterisk/config.h"
 
+#define AST_SIP_API_VERSION 1
+
+struct ast_sip_api_tech {
+	const int version;
+	const char *name;
+	int (*sipinfo_send)(struct ast_channel *chan,
+			struct ast_variable *headers, const char *content_type,
+			const char *content, const char *useragent_filter);
+};
+
 /*!
  * \brief Send a customized SIP INFO request
  *
@@ -44,6 +54,23 @@ int ast_sipinfo_send(struct ast_channel *chan,
 		const char *content,
 		const char *useragent_filter);
 
+/*!
+ * \brief Register a SIP API provider
+ *
+ * This will fail if a provider has already registered or if the
+ * provider is using an incorrect version.
+ *
+ * \param provider The provider to register
+ * \retval 0 Success
+ * \retval -1 Failure
+ */
+int ast_sip_api_provider_register(const struct ast_sip_api_tech *provider);
+
+/*!
+ * \brief Unregister a SIP API provider
+ */
+void ast_sip_api_provider_unregister(void);
+
 #if defined(__cplusplus) || defined(c_plusplus)
 }
 #endif

main/manager.c

@@ -4369,7 +4369,7 @@ static int action_originate(struct mansession *s, const struct message *m)
 
 	/* Allocate requested channel variables */
 	vars = astman_get_variables(m);
-	if (s->session->chanvars) {
+	if (s->session && s->session->chanvars) {
 		struct ast_variable *v, *old;
 		old = vars;
 		vars = NULL;

main/sip_api.c

@@ -0,0 +1,60 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2012, 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.
+ */
+#include "asterisk.h"
+
+#include "asterisk/sip_api.h"
+#include "asterisk/logger.h"
+
+static const struct ast_sip_api_tech *api_provider;
+
+int ast_sipinfo_send(struct ast_channel *chan,
+		struct ast_variable *headers,
+		const char *content_type,
+		const char *content,
+		const char *useragent_filter)
+{
+	if (!api_provider) {
+		ast_log(LOG_WARNING, "Unable to send custom SIP INFO. No API provider registered\n");
+		return -1;
+	}
+
+	return api_provider->sipinfo_send(chan, headers, content_type, content, useragent_filter);
+}
+
+int ast_sip_api_provider_register(const struct ast_sip_api_tech *provider)
+{
+	if (api_provider) {
+		ast_log(LOG_WARNING, "SIP provider %s has already registered. Not registering provider %s\n",
+				api_provider->name, provider->name);
+		return -1;
+	}
+
+	if (provider->version != AST_SIP_API_VERSION) {
+		ast_log(LOG_WARNING, "SIP API provider version mismatch: Current version is %d but provider "
+				"uses version %d\n", AST_SIP_API_VERSION, provider->version);
+		return -1;
+	}
+
+	api_provider = provider;
+	return 0;
+}
+
+void ast_sip_api_provider_unregister(void)
+{
+	api_provider = NULL;
+}

main/tcptls.c

@@ -192,11 +192,19 @@ static void *handle_tcptls_connection(void *data)
 				long res;
 				peer = SSL_get_peer_certificate(tcptls_session->ssl);
 				if (!peer) {
-					ast_log(LOG_WARNING, "No peer SSL certificate\n");
+					ast_log(LOG_ERROR, "No peer SSL certificate to verify\n");
+					ast_tcptls_close_session_file(tcptls_session);
+					ao2_ref(tcptls_session, -1);
+					return NULL;
 				}
+
 				res = SSL_get_verify_result(tcptls_session->ssl);
 				if (res != X509_V_OK) {
 					ast_log(LOG_ERROR, "Certificate did not verify: %s\n", X509_verify_cert_error_string(res));
+					X509_free(peer);
+					ast_tcptls_close_session_file(tcptls_session);
+					ao2_ref(tcptls_session, -1);
+					return NULL;
 				}
 				if (!ast_test_flag(&tcptls_session->parent->tls_cfg->flags, AST_SSL_IGNORE_COMMON_NAME)) {
 					ASN1_STRING *str;
@@ -227,17 +235,13 @@ static void *handle_tcptls_connection(void *data)
 					}
 					if (!found) {
 						ast_log(LOG_ERROR, "Certificate common name did not match (%s)\n", tcptls_session->parent->hostname);
-						if (peer) {
-							X509_free(peer);
-						}
+						X509_free(peer);
 						ast_tcptls_close_session_file(tcptls_session);
 						ao2_ref(tcptls_session, -1);
 						return NULL;
 					}
 				}
-				if (peer) {
-					X509_free(peer);
-				}
+				X509_free(peer);
 			}
 		}
 		if (!tcptls_session->f) {	/* no success opening descriptor stacking */