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	Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0).
		
			
				
	
	
		
			170 lines
		
	
	
		
			4.7 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			170 lines
		
	
	
		
			4.7 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
/*
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 * Asterisk -- An open source telephony toolkit.
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 *
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 * Copyright (C) 1999 - 2005, Digium, Inc.
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 *
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 * Mark Spencer <markster@digium.com>
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 *
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 * See http://www.asterisk.org for more information about
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 * the Asterisk project. Please do not directly contact
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 * any of the maintainers of this project for assistance;
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 * the project provides a web site, mailing lists and IRC
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 * channels for your use.
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 *
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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
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 * at the top of the source tree.
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 */
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/*! \file
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 *
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 * \brief Transfer a caller
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 *
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 * \author Mark Spencer <markster@digium.com>
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 *
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 * Requires transfer support from channel driver
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 *
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 * \ingroup applications
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 */
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/*** MODULEINFO
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	<support_level>core</support_level>
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 ***/
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#include "asterisk.h"
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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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/*** DOCUMENTATION
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	<application name="Transfer" language="en_US">
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		<since>
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			<version>0.5.0</version>
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		</since>
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		<synopsis>
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			Transfer caller to remote extension.
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		</synopsis>
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		<syntax>
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			<parameter name="dest" required="true" argsep="">
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				<argument name="Tech/" />
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				<argument name="destination" required="true" />
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			</parameter>
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		</syntax>
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		<description>
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			<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
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			SIP message will be returned to the caller.</para>
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			<para>The result of the application will be reported in the <variable>TRANSFERSTATUS</variable>
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			channel variable:</para>
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			<variablelist>
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				<variable name="TRANSFERSTATUS">
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					<value name="SUCCESS">
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						Transfer succeeded.
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					</value>
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					<value name="FAILURE">
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						Transfer failed.
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					</value>
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					<value name="UNSUPPORTED">
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						Transfer unsupported by channel driver.
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					</value>
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				</variable>
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				<variable name="TRANSFERSTATUSPROTOCOL">
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					<value name="0">
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						No error.
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					</value>
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					<value name="3xx-6xx">
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						SIP example - Error result code.
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					</value>
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				</variable>
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			</variablelist>
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		</description>
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	</application>
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 ***/
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static const char * const app = "Transfer";
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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;
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	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;
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	char status_protocol[20];
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	AST_DECLARE_APP_ARGS(args,
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		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);
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		pbx_builtin_setvar_helper(chan, "TRANSFERSTATUSPROTOCOL", status_protocol);
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		return 0;
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	} else
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		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))) {
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		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);
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			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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	/* 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);
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		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
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	   SIP example, 0 = success, 3xx-6xx are sip error codes for the REFER */
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	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 {
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		status = "SUCCESS";
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		res = 0;
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	}
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	snprintf(status_protocol, sizeof(status_protocol), "%d", protocol);
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	ast_debug(1, "ast_transfer channel %s TRANSFERSTATUS=%s, TRANSFERSTATUSPROTOCOL=%s\n",
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		  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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	return res;
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}
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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");
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