Files
asterisk/apps/app_transfer.c
George Joseph e6fe538d81 docs: Add version information to application and function XML elements
* Do a git blame on the embedded XML application or function element.

* From the commit hash, grab the summary line.

* Do a git log --grep <summary> to find the cherry-pick commits in all
  branches that match.

* Do a git patch-id to ensure the commits are all related and didn't get
  a false match on the summary.

* Do a git tag --contains <commit> to find the tags that contain each
  commit.

* Weed out all tags not ..0.

* Sort and discard any .0.0 and following tags where the commit
  appeared in an earlier branch.

* The result is a single tag for each branch where the application or function
  was defined.

The applications and functions defined in the following files were done by
hand because the XML was extracted from the C source file relatively recently.
* channels/pjsip/dialplan_functions_doc.xml
* main/logger_doc.xml
* main/manager_doc.xml
* res/res_geolocation/geoloc_doc.xml
* res/res_stir_shaken/stir_shaken_doc.xml

(cherry picked from commit 54d67711f8)
2025-01-23 18:36:04 +00:00

168 lines
4.7 KiB
C

/*
* 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
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/app.h"
#include "asterisk/channel.h"
/*** DOCUMENTATION
<application name="Transfer" language="en_US">
<since><version>1.6.2.0</version></since>
<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, etc) is used, only
an incoming call with the same channel technology will be transferred.
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>
<variable name="TRANSFERSTATUSPROTOCOL">
<value name="0">
No error.
</value>
<value name="3xx-6xx">
SIP example - Error result code.
</value>
</variable>
</variablelist>
</description>
</application>
***/
static const char * const app = "Transfer";
static int transfer_exec(struct ast_channel *chan, const char *data)
{
int res;
int len;
char *slash;
char *tech = NULL;
char *dest = NULL;
char *status;
char *parse;
int protocol = 0;
char status_protocol[20];
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");
snprintf(status_protocol, sizeof(status_protocol), "%d", protocol);
pbx_builtin_setvar_helper(chan, "TRANSFERSTATUSPROTOCOL", status_protocol);
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(ast_channel_tech(chan)->type, tech, len)) {
pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "FAILURE");
snprintf(status_protocol, sizeof(status_protocol), "%d", protocol);
pbx_builtin_setvar_helper(chan, "TRANSFERSTATUSPROTOCOL", status_protocol);
return 0;
}
}
/* Check if the channel supports transfer before we try it */
if (!ast_channel_tech(chan)->transfer) {
pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", "UNSUPPORTED");
snprintf(status_protocol, sizeof(status_protocol), "%d", protocol);
pbx_builtin_setvar_helper(chan, "TRANSFERSTATUSPROTOCOL", status_protocol);
return 0;
}
/* 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);
if (res < 0) {
status = "FAILURE";
res = 0;
} else {
status = "SUCCESS";
res = 0;
}
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);
pbx_builtin_setvar_helper(chan, "TRANSFERSTATUS", status);
pbx_builtin_setvar_helper(chan, "TRANSFERSTATUSPROTOCOL", status_protocol);
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");