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

1 files changed, 129 insertions, 0 deletions

diff --git a/res/res_pjsip_dtmf_info.c b/res/res_pjsip_dtmf_info.c
new file mode 100644
index 000000000..6c69b5318
--- /dev/null
+++ b/res/res_pjsip_dtmf_info.c
@@ -0,0 +1,129 @@
+/*
+ * 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>
+	<depend>res_pjsip</depend>
+	<depend>res_pjsip_session</depend>
+	<support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include <pjsip.h>
+#include <pjsip_ua.h>
+
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_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 (!ast_sip_is_content_type(&body->content_type, "application", "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, "PJSIP DTMF INFO Support",
+		.load = load_module,
+		.unload = unload_module,
+		.load_pri = AST_MODPRI_APP_DEPEND,
+	       );