Merge the pimp_my_sip branch into trunk.

The pimp_my_sip branch is being merged at this point because
it offers basic functionality, and from an API standpoint, things
are complete.

SIP work is *not* feature-complete; however, with the completion
of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have
been created, and thus it is possible for developers to attempt
to create new SIP work.

API documentation can be found in the doxygen in the code, but
usability documentation is still lacking.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3

res/res_sip_rfc3326.c

@@ -0,0 +1,145 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2013, Digium, Inc.
+ *
+ * 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>
+	<support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include <pjsip.h>
+#include <pjsip_ua.h>
+
+#include "asterisk/res_sip.h"
+#include "asterisk/res_sip_session.h"
+#include "asterisk/module.h"
+#include "asterisk/causes.h"
+
+static void rfc3326_use_reason_header(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
+{
+	const pj_str_t str_reason = { "Reason", 6 };
+	pjsip_generic_string_hdr *header = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_reason, NULL);
+	char buf[20], *cause, *text;
+	int code;
+
+	if (!header) {
+		return;
+	}
+
+	ast_copy_pj_str(buf, &header->hvalue, sizeof(buf));
+	cause = ast_skip_blanks(buf);
+
+	if (strncasecmp(cause, "Q.850", 5) || !(cause = strstr(cause, "cause="))) {
+		return;
+	}
+
+	/* If text is present get rid of it */
+	if ((text = strstr(cause, ";"))) {
+		*text = '\0';
+	}
+
+	if (sscanf(cause, "cause=%30d", &code) != 1) {
+		return;
+	}
+
+	ast_channel_hangupcause_set(session->channel, code & 0x7f);
+}
+
+static int rfc3326_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
+{
+	if ((pjsip_method_cmp(&rdata->msg_info.msg->line.req.method, &pjsip_bye_method) &&
+	     pjsip_method_cmp(&rdata->msg_info.msg->line.req.method, &pjsip_cancel_method)) ||
+	    !session->channel) {
+		return 0;
+	}
+
+	rfc3326_use_reason_header(session, rdata);
+
+	return 0;
+}
+
+static void rfc3326_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
+{
+	struct pjsip_status_line status = rdata->msg_info.msg->line.status;
+
+	if ((status.code < 300) || !session->channel) {
+		return;
+	}
+
+	rfc3326_use_reason_header(session, rdata);
+}
+
+static void rfc3326_add_reason_header(struct ast_sip_session *session, struct pjsip_tx_data *tdata)
+{
+	char buf[20];
+
+	snprintf(buf, sizeof(buf), "Q.850;cause=%i", ast_channel_hangupcause(session->channel) & 0x7f);
+	ast_sip_add_header(tdata, "Reason", buf);
+
+	if (ast_channel_hangupcause(session->channel) == AST_CAUSE_ANSWERED_ELSEWHERE) {
+		ast_sip_add_header(tdata, "Reason", "SIP;cause=200;text=\"Call completed elsewhere\"");
+	}
+}
+
+static void rfc3326_outgoing_request(struct ast_sip_session *session, struct pjsip_tx_data *tdata)
+{
+	if ((pjsip_method_cmp(&tdata->msg->line.req.method, &pjsip_bye_method) &&
+	     pjsip_method_cmp(&tdata->msg->line.req.method, &pjsip_cancel_method)) ||
+	    !session->channel) {
+		return;
+	}
+
+	rfc3326_add_reason_header(session, tdata);
+}
+
+static void rfc3326_outgoing_response(struct ast_sip_session *session, struct pjsip_tx_data *tdata)
+{
+	struct pjsip_status_line status = tdata->msg->line.status;
+
+	if ((status.code < 300) || !session->channel) {
+		return;
+	}
+
+	rfc3326_add_reason_header(session, tdata);
+}
+
+static struct ast_sip_session_supplement rfc3326_supplement = {
+	.incoming_request = rfc3326_incoming_request,
+	.incoming_response = rfc3326_incoming_response,
+	.outgoing_request = rfc3326_outgoing_request,
+	.outgoing_response = rfc3326_outgoing_response,
+};
+
+static int load_module(void)
+{
+	ast_sip_session_register_supplement(&rfc3326_supplement);
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	ast_sip_session_unregister_supplement(&rfc3326_supplement);
+	return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP RFC3326 Support",
+		.load = load_module,
+		.unload = unload_module,
+		.load_pri = AST_MODPRI_APP_DEPEND,
+	       );