the 'i' options for app_dial and app_queue, in that they will ignore any attempts by phones to forward the call. (closes issue #13977) Reported by: putnopvut Patches: page_ignore_forwards.patch uploaded by putnopvut (license 60) Tested by: putnopvut, acunningham git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
269 lines
7.5 KiB
C
269 lines
7.5 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (c) 2004 - 2006 Digium, Inc. All rights reserved.
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*
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* Mark Spencer <markster@digium.com>
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*
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* This code is released under the GNU General Public License
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* version 2.0. See LICENSE for more information.
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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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*/
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/*! \file
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*
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* \brief page() - Paging application
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*
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* \author Mark Spencer <markster@digium.com>
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*
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* \ingroup applications
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*/
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/*** MODULEINFO
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<depend>dahdi</depend>
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<depend>app_meetme</depend>
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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/channel.h"
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#include "asterisk/pbx.h"
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#include "asterisk/module.h"
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#include "asterisk/file.h"
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#include "asterisk/app.h"
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#include "asterisk/chanvars.h"
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#include "asterisk/utils.h"
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#include "asterisk/devicestate.h"
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#include "asterisk/dial.h"
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/*** DOCUMENTATION
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<application name="Page" language="en_US">
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<synopsis>
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Page series of phones
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</synopsis>
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<syntax>
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<parameter name="Technology/Resource" required="true" argsep="&">
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<argument name="Technology/Resource" required="true">
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<para>Specification of the device(s) to dial. These must be in the format of
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<literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
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represents a particular channel driver, and <replaceable>Resource</replaceable> represents a resource
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available to that particular channel driver.</para>
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</argument>
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<argument name="Technology2/Resource2" multiple="true">
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<para>Optional extra devices to dial inparallel</para>
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<para>If you need more then one enter them as Technology2/Resource2&
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Technology3/Resourse3&.....</para>
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</argument>
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</parameter>
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<parameter name="options">
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<optionlist>
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<option name="d">
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<para>Full duplex audio</para>
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</option>
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<option name="i">
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<para>Ignore attempts to forward the call</para>
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</option>
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<option name="q">
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<para>Quiet, do not play beep to caller</para>
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</option>
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<option name="r">
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<para>Record the page into a file (meetme option <literal>r</literal>)</para>
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</option>
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<option name="s">
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<para>Only dial a channel if its device state says that it is <literal>NOT_INUSE</literal></para>
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</option>
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</optionlist>
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</parameter>
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<parameter name="timeout">
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<para>Specify the length of time that the system will attempt to connect a call.
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After this duration, any intercom calls that have not been answered will be hung up by the
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system.</para>
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</parameter>
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</syntax>
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<description>
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<para>Places outbound calls to the given <replaceable>technology</replaceable>/<replaceable>resource</replaceable>
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and dumps them into a conference bridge as muted participants. The original
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caller is dumped into the conference as a speaker and the room is
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destroyed when the original callers leaves.</para>
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</description>
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<see-also>
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<ref type="application">MeetMe</ref>
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</see-also>
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</application>
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***/
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static const char *app_page= "Page";
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enum {
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PAGE_DUPLEX = (1 << 0),
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PAGE_QUIET = (1 << 1),
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PAGE_RECORD = (1 << 2),
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PAGE_SKIP = (1 << 3),
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PAGE_IGNORE_FORWARDS = (1 << 4),
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} page_opt_flags;
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AST_APP_OPTIONS(page_opts, {
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AST_APP_OPTION('d', PAGE_DUPLEX),
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AST_APP_OPTION('q', PAGE_QUIET),
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AST_APP_OPTION('r', PAGE_RECORD),
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AST_APP_OPTION('s', PAGE_SKIP),
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AST_APP_OPTION('i', PAGE_IGNORE_FORWARDS),
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});
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#define MAX_DIALS 128
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static int page_exec(struct ast_channel *chan, void *data)
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{
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char *tech, *resource, *tmp;
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char meetmeopts[88], originator[AST_CHANNEL_NAME], *opts[0];
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struct ast_flags flags = { 0 };
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unsigned int confid = ast_random();
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struct ast_app *app;
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int res = 0, pos = 0, i = 0;
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struct ast_dial *dials[MAX_DIALS];
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int timeout = 0;
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char *parse;
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AST_DECLARE_APP_ARGS(args,
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AST_APP_ARG(devices);
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AST_APP_ARG(options);
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AST_APP_ARG(timeout);
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);
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if (ast_strlen_zero(data)) {
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ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
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return -1;
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}
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if (!(app = pbx_findapp("MeetMe"))) {
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ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
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return -1;
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};
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parse = ast_strdupa(data);
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AST_STANDARD_APP_ARGS(args, parse);
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ast_copy_string(originator, chan->name, sizeof(originator));
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if ((tmp = strchr(originator, '-'))) {
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*tmp = '\0';
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}
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if (!ast_strlen_zero(args.options)) {
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ast_app_parse_options(page_opts, &flags, opts, args.options);
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}
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if (!ast_strlen_zero(args.timeout)) {
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timeout = atoi(args.timeout);
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}
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snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
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(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
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/* Go through parsing/calling each device */
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while ((tech = strsep(&args.devices, "&"))) {
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int state = 0;
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struct ast_dial *dial = NULL;
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/* don't call the originating device */
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if (!strcasecmp(tech, originator))
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continue;
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/* If no resource is available, continue on */
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if (!(resource = strchr(tech, '/'))) {
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ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
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continue;
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}
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/* Ensure device is not in use if skip option is enabled */
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if (ast_test_flag(&flags, PAGE_SKIP)) {
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state = ast_device_state(tech);
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if (state == AST_DEVICE_UNKNOWN) {
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ast_log(LOG_WARNING, "Destination '%s' has device state '%s'. Paging anyway.\n", tech, ast_devstate2str(state));
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} else if (state != AST_DEVICE_NOT_INUSE) {
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ast_log(LOG_WARNING, "Destination '%s' has device state '%s'.\n", tech, ast_devstate2str(state));
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continue;
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}
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}
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*resource++ = '\0';
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/* Create a dialing structure */
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if (!(dial = ast_dial_create())) {
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ast_log(LOG_WARNING, "Failed to create dialing structure.\n");
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continue;
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}
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/* Append technology and resource */
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if (ast_dial_append(dial, tech, resource) == -1) {
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ast_log(LOG_ERROR, "Failed to add %s to outbound dial\n", tech);
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continue;
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}
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/* Set ANSWER_EXEC as global option */
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ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
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if (timeout) {
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ast_dial_set_global_timeout(dial, timeout * 1000);
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}
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if (ast_test_flag(&flags, PAGE_IGNORE_FORWARDS)) {
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ast_dial_option_global_enable(dial, AST_DIAL_OPTION_DISABLE_CALL_FORWARDING, NULL);
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}
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/* Run this dial in async mode */
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ast_dial_run(dial, chan, 1);
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/* Put in our dialing array */
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dials[pos++] = dial;
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}
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if (!ast_test_flag(&flags, PAGE_QUIET)) {
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res = ast_streamfile(chan, "beep", chan->language);
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if (!res)
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res = ast_waitstream(chan, "");
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}
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if (!res) {
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snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"),
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(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
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pbx_exec(chan, app, meetmeopts);
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}
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/* Go through each dial attempt cancelling, joining, and destroying */
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for (i = 0; i < pos; i++) {
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struct ast_dial *dial = dials[i];
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/* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
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ast_dial_join(dial);
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/* Hangup all channels */
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ast_dial_hangup(dial);
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/* Destroy dialing structure */
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ast_dial_destroy(dial);
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}
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return -1;
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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_page);
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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_page, page_exec);
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");
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