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, 147 insertions, 0 deletions

diff --git a/res/res_pjsip_rfc3326.c b/res/res_pjsip_rfc3326.c
new file mode 100644
index 000000000..66594fef5
--- /dev/null
+++ b/res/res_pjsip_rfc3326.c
@@ -0,0 +1,147 @@
+/*
+ * 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>
+	<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"
+#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, "PJSIP RFC3326 Support",
+		.load = load_module,
+		.unload = unload_module,
+		.load_pri = AST_MODPRI_APP_DEPEND,
+	       );