This commit introduces the first phase of an effort to manage documentation of the interfaces in Asterisk in an XML format. Currently, a new format is available for applications and dialplan functions. A good number of conversions to the new format are also included. For more information, see the following message to asterisk-dev: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
143 lines
3.5 KiB
C
143 lines
3.5 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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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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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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<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, LOCAL etc) is used, only
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an incoming call with the same channel technology will be transfered.
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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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</variablelist>
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</description>
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</application>
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***/
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static const char *app = "Transfer";
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static int transfer_exec(struct ast_channel *chan, void *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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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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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(chan->tech->type, tech, len)) {
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pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "FAILURE");
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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 (!chan->tech->transfer) {
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pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "UNSUPPORTED");
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return 0;
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}
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res = ast_transfer(chan, dest);
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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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pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", status);
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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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