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, 0 insertions, 68 deletions

diff --git a/res/res_sip_endpoint_identifier_constant.c b/res/res_sip_endpoint_identifier_constant.c
deleted file mode 100644
index 212cca263..000000000
--- a/res/res_sip_endpoint_identifier_constant.c
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 2013, Digium, Inc.
- *
- * Mark Michelson <mmichelson@digium.com>
- *
- * Includes code and algorithms from the Zapata library.
- *
- * 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_sip</depend>
-	<defaultenabled>no</defaultenabled>
-	<support_level>core</support_level>
- ***/
-
-#include "asterisk.h"
-
-#include <pjsip.h>
-
-#include "asterisk/res_sip.h"
-#include "asterisk/module.h"
-
-static struct ast_sip_endpoint *constant_identify(pjsip_rx_data *rdata)
-{
-	/* This endpoint identifier always returns the same endpoint. It's used
-	 * simply for testing. It allocates an endpoint from sorcery so default values
-	 * do get applied.
-	 */
-	struct ast_sip_endpoint *endpoint = ast_sorcery_alloc(ast_sip_get_sorcery(), "endpoint", NULL);
-	if (!endpoint) {
-		return NULL;
-	}
-	ast_parse_allow_disallow(&endpoint->prefs, endpoint->codecs, "ulaw", 1);
-	return endpoint;
-}
-
-static struct ast_sip_endpoint_identifier constant_identifier = {
-	.identify_endpoint = constant_identify,
-};
-
-static int load_module(void)
-{
-	ast_sip_register_endpoint_identifier(&constant_identifier);
-	return AST_MODULE_LOAD_SUCCESS;
-}
-
-static int unload_module(void)
-{
-	return 0;
-}
-
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP Constant Endpoint Identifier",
-		.load = load_module,
-		.unload = unload_module,
-		.load_pri = AST_MODPRI_APP_DEPEND,
-	       );