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, 82 insertions, 0 deletions

diff --git a/res/res_pjsip_logger.c b/res/res_pjsip_logger.c
new file mode 100644
index 000000000..a013bb5a5
--- /dev/null
+++ b/res/res_pjsip_logger.c
@@ -0,0 +1,82 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2013, Digium, Inc.
+ *
+ * Mark Michelson <mmichelson@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>
+	<support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include <pjsip.h>
+
+#include "asterisk/res_pjsip.h"
+#include "asterisk/module.h"
+#include "asterisk/logger.h"
+
+static pj_status_t logging_on_tx_msg(pjsip_tx_data *tdata)
+{
+	ast_verbose("<--- Transmitting SIP %s (%d bytes) to %s:%s:%d --->\n%.*s\n",
+		    tdata->msg->type == PJSIP_REQUEST_MSG ? "request" : "response",
+		    (int) (tdata->buf.cur - tdata->buf.start),
+		    tdata->tp_info.transport->type_name,
+		    tdata->tp_info.dst_name,
+		    tdata->tp_info.dst_port,
+		    (int) (tdata->buf.end - tdata->buf.start), tdata->buf.start);
+	return PJ_SUCCESS;
+}
+
+static pj_bool_t logging_on_rx_msg(pjsip_rx_data *rdata)
+{
+	ast_verbose("<--- Received SIP %s (%d bytes) from %s:%s:%d --->\n%s\n",
+		    rdata->msg_info.msg->type == PJSIP_REQUEST_MSG ? "request" : "response",
+		    rdata->msg_info.len,
+		    rdata->tp_info.transport->type_name,
+		    rdata->pkt_info.src_name,
+		    rdata->pkt_info.src_port,
+		    rdata->pkt_info.packet);
+	return PJ_FALSE;
+}
+
+static pjsip_module logging_module = {
+	.name = { "Logging Module", 14 },
+	.priority = 0,
+	.on_rx_request = logging_on_rx_msg,
+	.on_rx_response = logging_on_rx_msg,
+	.on_tx_request = logging_on_tx_msg,
+	.on_tx_response = logging_on_tx_msg,
+};
+
+static int load_module(void)
+{
+	ast_sip_register_service(&logging_module);
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+static int unload_module(void)
+{
+	ast_sip_unregister_service(&logging_module);
+	return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Packet Logger",
+		.load = load_module,
+		.unload = unload_module,
+		.load_pri = AST_MODPRI_APP_DEPEND,
+	       );