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										 |  |  | /*
 | 
					
						
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										 |  |  |  * Asterisk -- An open source telephony toolkit. | 
					
						
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										 |  |  |  * | 
					
						
							| 
									
										
										
										
											2005-01-21 07:06:25 +00:00
										 |  |  |  * Copyright (C) 1999 - 2005, Digium, Inc. | 
					
						
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										 |  |  |  * | 
					
						
							| 
									
										
										
										
											2004-10-02 00:58:31 +00:00
										 |  |  |  * Mark Spencer <markster@digium.com> | 
					
						
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										 |  |  |  * | 
					
						
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										 |  |  |  * 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. | 
					
						
							|  |  |  |  * | 
					
						
							| 
									
										
										
										
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										 |  |  |  * This program is free software, distributed under the terms of | 
					
						
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										 |  |  |  * the GNU General Public License Version 2. See the LICENSE file | 
					
						
							|  |  |  |  * at the top of the source tree. | 
					
						
							|  |  |  |  */ | 
					
						
							|  |  |  | 
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											2005-10-24 20:12:06 +00:00
										 |  |  | /*! \file
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										 |  |  |  * | 
					
						
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										 |  |  |  * \brief Transfer a caller | 
					
						
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											2005-12-30 21:18:06 +00:00
										 |  |  |  * | 
					
						
							|  |  |  |  * \author Mark Spencer <markster@digium.com> | 
					
						
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										 |  |  |  * | 
					
						
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											2005-11-06 15:09:47 +00:00
										 |  |  |  * Requires transfer support from channel driver | 
					
						
							|  |  |  |  * | 
					
						
							|  |  |  |  * \ingroup applications | 
					
						
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										 |  |  |  */ | 
					
						
							|  |  |  | 
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											2011-07-14 20:28:54 +00:00
										 |  |  | /*** MODULEINFO
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							|  |  |  | 	<support_level>core</support_level> | 
					
						
							|  |  |  |  ***/ | 
					
						
							|  |  |  | 
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										 |  |  | #include "asterisk.h"
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							|  |  |  | 
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										 |  |  | #include "asterisk/pbx.h"
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							|  |  |  | #include "asterisk/module.h"
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										 |  |  | #include "asterisk/app.h"
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										 |  |  | #include "asterisk/channel.h"
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										 |  |  | 
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										 |  |  | /*** DOCUMENTATION
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							|  |  |  | 	<application name="Transfer" language="en_US"> | 
					
						
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										 |  |  | 		<since> | 
					
						
							|  |  |  | 			<version>0.5.0</version> | 
					
						
							|  |  |  | 		</since> | 
					
						
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										 |  |  | 		<synopsis> | 
					
						
							|  |  |  | 			Transfer caller to remote extension. | 
					
						
							|  |  |  | 		</synopsis> | 
					
						
							|  |  |  | 		<syntax> | 
					
						
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										 |  |  | 			<parameter name="dest" required="true" argsep=""> | 
					
						
							|  |  |  | 				<argument name="Tech/" /> | 
					
						
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										 |  |  | 				<argument name="destination" required="true" /> | 
					
						
							|  |  |  | 			</parameter> | 
					
						
							|  |  |  | 		</syntax> | 
					
						
							|  |  |  | 		<description> | 
					
						
							|  |  |  | 			<para>Requests the remote caller be transferred | 
					
						
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										 |  |  | 			to a given destination. If TECH (SIP, IAX2, etc) is used, only | 
					
						
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										 |  |  | 			an incoming call with the same channel technology will be transferred. | 
					
						
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										 |  |  | 			Note that for SIP, if you transfer before call is setup, a 302 redirect | 
					
						
							|  |  |  | 			SIP message will be returned to the caller.</para> | 
					
						
							|  |  |  | 			<para>The result of the application will be reported in the <variable>TRANSFERSTATUS</variable> | 
					
						
							|  |  |  | 			channel variable:</para> | 
					
						
							|  |  |  | 			<variablelist> | 
					
						
							|  |  |  | 				<variable name="TRANSFERSTATUS"> | 
					
						
							|  |  |  | 					<value name="SUCCESS"> | 
					
						
							|  |  |  | 						Transfer succeeded. | 
					
						
							|  |  |  | 					</value> | 
					
						
							|  |  |  | 					<value name="FAILURE"> | 
					
						
							|  |  |  | 						Transfer failed. | 
					
						
							|  |  |  | 					</value> | 
					
						
							|  |  |  | 					<value name="UNSUPPORTED"> | 
					
						
							|  |  |  | 						Transfer unsupported by channel driver. | 
					
						
							|  |  |  | 					</value> | 
					
						
							|  |  |  | 				</variable> | 
					
						
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										 |  |  | 				<variable name="TRANSFERSTATUSPROTOCOL"> | 
					
						
							|  |  |  | 					<value name="0"> | 
					
						
							|  |  |  | 						No error. | 
					
						
							|  |  |  | 					</value> | 
					
						
							|  |  |  | 					<value name="3xx-6xx"> | 
					
						
							|  |  |  | 						SIP example - Error result code. | 
					
						
							|  |  |  | 					</value> | 
					
						
							|  |  |  | 				</variable> | 
					
						
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										 |  |  | 			</variablelist> | 
					
						
							|  |  |  | 		</description> | 
					
						
							|  |  |  | 	</application> | 
					
						
							|  |  |  |  ***/ | 
					
						
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										 |  |  | 
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										 |  |  | static const char * const app = "Transfer"; | 
					
						
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										 |  |  | 
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										 |  |  | static int transfer_exec(struct ast_channel *chan, const char *data) | 
					
						
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										 |  |  | { | 
					
						
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										 |  |  | 	int res; | 
					
						
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										 |  |  | 	int len; | 
					
						
							|  |  |  | 	char *slash; | 
					
						
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										 |  |  | 	char *tech = NULL; | 
					
						
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										 |  |  | 	char *dest = NULL; | 
					
						
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										 |  |  | 	char *status; | 
					
						
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										 |  |  | 	char *parse; | 
					
						
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										 |  |  | 	int protocol = 0; | 
					
						
							|  |  |  | 	char status_protocol[20]; | 
					
						
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										 |  |  | 	AST_DECLARE_APP_ARGS(args, | 
					
						
							|  |  |  | 		AST_APP_ARG(dest); | 
					
						
							|  |  |  | 	); | 
					
						
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										 |  |  | 
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										 |  |  | 	if (ast_strlen_zero((char *)data)) { | 
					
						
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										 |  |  | 		ast_log(LOG_WARNING, "Transfer requires an argument ([Tech/]destination)\n"); | 
					
						
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										 |  |  | 		pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "FAILURE"); | 
					
						
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										 |  |  | 		snprintf(status_protocol, sizeof(status_protocol), "%d", protocol); | 
					
						
							|  |  |  | 		pbx_builtin_setvar_helper(chan, "TRANSFERSTATUSPROTOCOL", status_protocol); | 
					
						
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										 |  |  | 		return 0; | 
					
						
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										 |  |  | 	} else | 
					
						
							|  |  |  | 		parse = ast_strdupa(data); | 
					
						
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										 |  |  | 	AST_STANDARD_APP_ARGS(args, parse); | 
					
						
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							|  |  |  | 	dest = args.dest; | 
					
						
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										 |  |  | 	if ((slash = strchr(dest, '/')) && (len = (slash - dest))) { | 
					
						
							|  |  |  | 		tech = dest; | 
					
						
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										 |  |  | 		dest = slash + 1; | 
					
						
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										 |  |  | 		/* Allow execution only if the Tech/destination agrees with the type of the channel */ | 
					
						
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										 |  |  | 		if (strncasecmp(ast_channel_tech(chan)->type, tech, len)) { | 
					
						
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										 |  |  | 			pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "FAILURE"); | 
					
						
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										 |  |  | 			snprintf(status_protocol, sizeof(status_protocol), "%d", protocol); | 
					
						
							|  |  |  | 			pbx_builtin_setvar_helper(chan, "TRANSFERSTATUSPROTOCOL", status_protocol); | 
					
						
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										 |  |  | 			return 0; | 
					
						
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										 |  |  | 		} | 
					
						
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										 |  |  | 	} | 
					
						
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										 |  |  | 
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										 |  |  | 	/* Check if the channel supports transfer before we try it */ | 
					
						
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										 |  |  | 	if (!ast_channel_tech(chan)->transfer) { | 
					
						
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										 |  |  | 		pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "UNSUPPORTED"); | 
					
						
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										 |  |  | 		snprintf(status_protocol, sizeof(status_protocol), "%d", protocol); | 
					
						
							|  |  |  | 		pbx_builtin_setvar_helper(chan, "TRANSFERSTATUSPROTOCOL", status_protocol); | 
					
						
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										 |  |  | 		return 0; | 
					
						
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										 |  |  | 	} | 
					
						
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										 |  |  | 	/* New transfer API returns a protocol code
 | 
					
						
							|  |  |  | 	   SIP example, 0 = success, 3xx-6xx are sip error codes for the REFER */ | 
					
						
							|  |  |  | 	res = ast_transfer_protocol(chan, dest, &protocol); | 
					
						
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										 |  |  | 	if (res < 0) { | 
					
						
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										 |  |  | 		status = "FAILURE"; | 
					
						
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										 |  |  | 		res = 0; | 
					
						
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										 |  |  | 	} else { | 
					
						
							|  |  |  | 		status = "SUCCESS"; | 
					
						
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										 |  |  | 		res = 0; | 
					
						
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										 |  |  | 	} | 
					
						
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										 |  |  | 	snprintf(status_protocol, sizeof(status_protocol), "%d", protocol); | 
					
						
							|  |  |  | 	ast_debug(1, "ast_transfer channel %s TRANSFERSTATUS=%s, TRANSFERSTATUSPROTOCOL=%s\n", | 
					
						
							|  |  |  | 		  ast_channel_name(chan), status, status_protocol); | 
					
						
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										 |  |  | 	pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", status); | 
					
						
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										 |  |  | 	pbx_builtin_setvar_helper(chan, "TRANSFERSTATUSPROTOCOL", status_protocol); | 
					
						
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										 |  |  | 
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										 |  |  | 	return res; | 
					
						
							|  |  |  | } | 
					
						
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										 |  |  | static int unload_module(void) | 
					
						
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										 |  |  | { | 
					
						
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										 |  |  | 	return ast_unregister_application(app); | 
					
						
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										 |  |  | } | 
					
						
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										 |  |  | static int load_module(void) | 
					
						
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										 |  |  | { | 
					
						
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										 |  |  | 	return ast_register_application_xml(app, transfer_exec); | 
					
						
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										 |  |  | } | 
					
						
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										 |  |  | AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Transfers a caller to another extension"); |