aboutsummaryrefslogtreecommitdiffstats
path: root/apps/app_transfer.c
blob: e1452bb5905e57c7fcb5680358f9a43dc48f0e48 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@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.
 */

/*! \file
 *
 * \brief Transfer a caller
 *
 * \author Mark Spencer <markster@digium.com>
 * 
 * Requires transfer support from channel driver
 *
 * \ingroup applications
 */

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/app.h"
#include "asterisk/channel.h"

/*** DOCUMENTATION
	<application name="Transfer" language="en_US">
		<synopsis>
			Transfer caller to remote extension.
		</synopsis>
		<syntax>
			<parameter name="dest" required="true" argsep="/">
				<argument name="Tech" />
				<argument name="destination" required="true" />
			</parameter>
		</syntax>
		<description>
			<para>Requests the remote caller be transferred
			to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only
			an incoming call with the same channel technology will be transfered.
			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>
			</variablelist>
		</description>
	</application>
 ***/

static const char *app = "Transfer";

static int transfer_exec(struct ast_channel *chan, void *data)
{
	int res;
	int len;
	char *slash;
	char *tech = NULL;
	char *dest = NULL;
	char *status;
	char *parse;
	AST_DECLARE_APP_ARGS(args,
		AST_APP_ARG(dest);
	);

	if (ast_strlen_zero((char *)data)) {
		ast_log(LOG_WARNING, "Transfer requires an argument ([Tech/]destination)\n");
		pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "FAILURE");
		return 0;
	} else
		parse = ast_strdupa(data);

	AST_STANDARD_APP_ARGS(args, parse);

	dest = args.dest;

	if ((slash = strchr(dest, '/')) && (len = (slash - dest))) {
		tech = dest;
		dest = slash + 1;
		/* Allow execution only if the Tech/destination agrees with the type of the channel */
		if (strncasecmp(chan->tech->type, tech, len)) {
			pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "FAILURE");
			return 0;
		}
	}

	/* Check if the channel supports transfer before we try it */
	if (!chan->tech->transfer) {
		pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "UNSUPPORTED");
		return 0;
	}

	res = ast_transfer(chan, dest);

	if (res < 0) {
		status = "FAILURE";
		res = 0;
	} else {
		status = "SUCCESS";
		res = 0;
	}

	pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", status);

	return res;
}

static int unload_module(void)
{
	return ast_unregister_application(app);
}

static int load_module(void)
{
	return ast_register_application_xml(app, transfer_exec);
}

AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Transfers a caller to another extension");