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

1 files changed, 128 insertions, 0 deletions

diff --git a/res/res_sip_dtmf_info.c b/res/res_sip_dtmf_info.c
new file mode 100644
index 000000000..453e57d06
--- /dev/null
+++ b/res/res_sip_dtmf_info.c
@@ -0,0 +1,128 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2013, Digium, Inc.
+ *
+ * Jason Parker <jparker@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"
+
+static int dtmf_info_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
+{
+	int res = 0;
+	pjsip_msg_body *body = rdata->msg_info.msg->body;
+
+	pjsip_tx_data *tdata;
+
+	char buf[body->len];
+	char *cur = buf;
+	char *line;
+
+	char event = '\0';
+	unsigned int duration = 0;
+
+	if (pj_strcmp2(&body->content_type.type, "application") ||
+	    pj_strcmp2(&body->content_type.subtype, "dtmf-relay")) {
+		return 0;
+	}
+
+	body->print_body(body, buf, body->len);
+
+	while ((line = strsep(&cur, "\r\n"))) {
+		char *c;
+
+		if (!(c = strchr(line, '='))) {
+			continue;
+		}
+		*c++ = '\0';
+
+		c = ast_skip_blanks(c);
+
+		if (!strcasecmp(line, "signal")) {
+			if (c[0] == '!' || c[0] == '*' || c[0] == '#' ||
+			    ('0' <= c[0] && c[0] <= '9') ||
+			    ('A' <= c[0] && c[0] <= 'D') ||
+			    ('a' <= c[0] && c[0] <= 'd')) {
+				event = c[0];
+			} else {
+				ast_log(LOG_ERROR, "Invalid DTMF event signal in INFO message.\n");
+				res = -1;
+				break;
+			}
+		} else if (!strcasecmp(line, "duration")) {
+			sscanf(c, "%30u", &duration);
+		}
+	}
+
+	if (!duration) {
+		duration = 100;
+	}
+
+	if (event == '!') {
+		struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_FLASH, } };
+
+		ast_queue_frame(session->channel, &f);
+	} else if (event != '\0') {
+		struct ast_frame f = { AST_FRAME_DTMF, };
+		f.len = duration;
+		f.subclass.integer = event;
+
+		ast_queue_frame(session->channel, &f);
+	} else {
+		res = -1;
+	}
+
+	if (pjsip_dlg_create_response(session->inv_session->dlg, rdata, !res ? 200 : 500, NULL, &tdata) == PJ_SUCCESS) {
+		struct pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
+
+		pjsip_dlg_send_response(session->inv_session->dlg, tsx, tdata);
+	}
+
+	return res;
+}
+
+static struct ast_sip_session_supplement dtmf_info_supplement = {
+	.method = "INFO",
+	.incoming_request = dtmf_info_incoming_request,
+};
+
+static int load_module(void)
+{
+	ast_sip_session_register_supplement(&dtmf_info_supplement);
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	ast_sip_session_unregister_supplement(&dtmf_info_supplement);
+	return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP DTMF INFO Support",
+		.load = load_module,
+		.unload = unload_module,
+		.load_pri = AST_MODPRI_APP_DEPEND,
+	       );