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, 128 insertions, 0 deletions

diff --git a/res/res_pjsip_one_touch_record_info.c b/res/res_pjsip_one_touch_record_info.c
new file mode 100644
index 000000000..f0ecbbfbf
--- /dev/null
+++ b/res/res_pjsip_one_touch_record_info.c
@@ -0,0 +1,128 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2013, malleable, llc.
+ *
+ * Sean Bright <sean@malleable.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/features.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
+#include "asterisk/module.h"
+#include "asterisk/features_config.h"
+
+static void send_response(struct ast_sip_session *session, int code, struct pjsip_rx_data *rdata)
+{
+	pjsip_tx_data *tdata;
+
+	if (pjsip_dlg_create_response(session->inv_session->dlg, rdata, code, NULL, &tdata) == PJ_SUCCESS) {
+		struct pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
+
+		pjsip_dlg_send_response(session->inv_session->dlg, tsx, tdata);
+	}
+}
+
+static int handle_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
+{
+	static const pj_str_t rec_str = { "Record", 6 };
+	pjsip_generic_string_hdr *record;
+	int feature_res;
+	char feature_code[AST_FEATURE_MAX_LEN];
+	const char *feature;
+	char *digit;
+
+	record = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &rec_str, NULL);
+
+	/* If we don't have Record header, we have nothing to do */
+	if (!record) {
+		return 0;
+	}
+
+	if (!pj_stricmp2(&record->hvalue, "on")) {
+		feature = session->endpoint->info.recording.onfeature;
+	} else if (!pj_stricmp2(&record->hvalue, "off")) {
+		feature = session->endpoint->info.recording.offfeature;
+	} else {
+		/* Don't send response because another module may handle this */
+		return 0;
+	}
+
+	if (!session->channel) {
+		send_response(session, 481, rdata);
+		return 0;
+	}
+
+	/* Is this endpoint configured with One Touch Recording? */
+	if (!session->endpoint->info.recording.enabled || ast_strlen_zero(feature)) {
+		send_response(session, 403, rdata);
+		return 0;
+	}
+
+	ast_channel_lock(session->channel);
+	feature_res = ast_get_feature(session->channel, feature, feature_code, sizeof(feature_code));
+	ast_channel_unlock(session->channel);
+
+	if (feature_res || ast_strlen_zero(feature_code)) {
+		send_response(session, 403, rdata);
+		return 0;
+	}
+
+	for (digit = feature_code; *digit; ++digit) {
+		struct ast_frame f = { AST_FRAME_DTMF, .subclass.integer = *digit, .len = 100 };
+		ast_queue_frame(session->channel, &f);
+	}
+
+	send_response(session, 200, rdata);
+
+	return 0;
+}
+
+static struct ast_sip_session_supplement info_supplement = {
+	.method = "INFO",
+	.incoming_request = handle_incoming_request,
+};
+
+static int load_module(void)
+{
+	if (ast_sip_session_register_supplement(&info_supplement)) {
+		ast_log(LOG_ERROR, "Unable to register One Touch Recording supplement\n");
+		return AST_MODULE_LOAD_FAILURE;
+	}
+
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	ast_sip_session_unregister_supplement(&info_supplement);
+	return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP INFO One Touch Recording Support",
+	.load = load_module,
+	.unload = unload_module,
+	.load_pri = AST_MODPRI_APP_DEPEND,
+	);