2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (c) 2004 - 2006 Digium, Inc. All rights reserved.
6 * Mark Spencer <markster@digium.com>
8 * This code is released under the GNU General Public License
9 * version 2.0. See LICENSE for more information.
11 * See http://www.asterisk.org for more information about
12 * the Asterisk project. Please do not directly contact
13 * any of the maintainers of this project for assistance;
14 * the project provides a web site, mailing lists and IRC
15 * channels for your use.
21 * \brief page() - Paging application
23 * \author Mark Spencer <markster@digium.com>
25 * \ingroup applications
29 <depend>dahdi</depend>
30 <depend>app_meetme</depend>
35 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
37 #include "asterisk/channel.h"
38 #include "asterisk/pbx.h"
39 #include "asterisk/module.h"
40 #include "asterisk/file.h"
41 #include "asterisk/app.h"
42 #include "asterisk/chanvars.h"
43 #include "asterisk/utils.h"
44 #include "asterisk/devicestate.h"
45 #include "asterisk/dial.h"
48 <application name="Page" language="en_US">
53 <parameter name="Technology/Resource" required="true" argsep="&">
54 <argument name="Technology/Resource" required="true">
55 <para>Specification of the device(s) to dial. These must be in the format of
56 <literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
57 represents a particular channel driver, and <replaceable>Resource</replaceable> represents a resource
58 available to that particular channel driver.</para>
60 <argument name="Technology2/Resource2" multiple="true">
61 <para>Optional extra devices to dial inparallel</para>
62 <para>If you need more then one enter them as Technology2/Resource2&
63 Technology3/Resourse3&.....</para>
66 <parameter name="options">
69 <para>Full duplex audio</para>
72 <para>Ignore attempts to forward the call</para>
75 <para>Quiet, do not play beep to caller</para>
78 <para>Record the page into a file (meetme option <literal>r</literal>)</para>
81 <para>Only dial a channel if its device state says that it is <literal>NOT_INUSE</literal></para>
85 <parameter name="timeout">
86 <para>Specify the length of time that the system will attempt to connect a call.
87 After this duration, any intercom calls that have not been answered will be hung up by the
92 <para>Places outbound calls to the given <replaceable>technology</replaceable>/<replaceable>resource</replaceable>
93 and dumps them into a conference bridge as muted participants. The original
94 caller is dumped into the conference as a speaker and the room is
95 destroyed when the original callers leaves.</para>
98 <ref type="application">MeetMe</ref>
102 static const char *app_page= "Page";
105 PAGE_DUPLEX = (1 << 0),
106 PAGE_QUIET = (1 << 1),
107 PAGE_RECORD = (1 << 2),
108 PAGE_SKIP = (1 << 3),
109 PAGE_IGNORE_FORWARDS = (1 << 4),
112 AST_APP_OPTIONS(page_opts, {
113 AST_APP_OPTION('d', PAGE_DUPLEX),
114 AST_APP_OPTION('q', PAGE_QUIET),
115 AST_APP_OPTION('r', PAGE_RECORD),
116 AST_APP_OPTION('s', PAGE_SKIP),
117 AST_APP_OPTION('i', PAGE_IGNORE_FORWARDS),
121 static int page_exec(struct ast_channel *chan, void *data)
123 char *tech, *resource, *tmp;
124 char meetmeopts[88], originator[AST_CHANNEL_NAME], *opts[0];
125 struct ast_flags flags = { 0 };
126 unsigned int confid = ast_random();
128 int res = 0, pos = 0, i = 0;
129 struct ast_dial **dial_list;
130 unsigned int num_dials;
134 AST_DECLARE_APP_ARGS(args,
135 AST_APP_ARG(devices);
136 AST_APP_ARG(options);
137 AST_APP_ARG(timeout);
140 if (ast_strlen_zero(data)) {
141 ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n");
145 if (!(app = pbx_findapp("MeetMe"))) {
146 ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
150 parse = ast_strdupa(data);
152 AST_STANDARD_APP_ARGS(args, parse);
154 ast_copy_string(originator, chan->name, sizeof(originator));
155 if ((tmp = strchr(originator, '-'))) {
159 if (!ast_strlen_zero(args.options)) {
160 ast_app_parse_options(page_opts, &flags, opts, args.options);
163 if (!ast_strlen_zero(args.timeout)) {
164 timeout = atoi(args.timeout);
167 snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
168 (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
170 /* Count number of extensions in list by number of ampersands + 1 */
180 if (!(dial_list = ast_calloc(num_dials, sizeof(struct ast_dial *)))) {
181 ast_log(LOG_ERROR, "Can't allocate %ld bytes for dial list\n", (long)(sizeof(struct ast_dial *) * num_dials));
185 /* Go through parsing/calling each device */
186 while ((tech = strsep(&args.devices, "&"))) {
188 struct ast_dial *dial = NULL;
190 /* don't call the originating device */
191 if (!strcasecmp(tech, originator))
194 /* If no resource is available, continue on */
195 if (!(resource = strchr(tech, '/'))) {
196 ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech);
200 /* Ensure device is not in use if skip option is enabled */
201 if (ast_test_flag(&flags, PAGE_SKIP)) {
202 state = ast_device_state(tech);
203 if (state == AST_DEVICE_UNKNOWN) {
204 ast_log(LOG_WARNING, "Destination '%s' has device state '%s'. Paging anyway.\n", tech, ast_devstate2str(state));
205 } else if (state != AST_DEVICE_NOT_INUSE) {
206 ast_log(LOG_WARNING, "Destination '%s' has device state '%s'.\n", tech, ast_devstate2str(state));
213 /* Create a dialing structure */
214 if (!(dial = ast_dial_create())) {
215 ast_log(LOG_WARNING, "Failed to create dialing structure.\n");
219 /* Append technology and resource */
220 if (ast_dial_append(dial, tech, resource) == -1) {
221 ast_log(LOG_ERROR, "Failed to add %s to outbound dial\n", tech);
225 /* Set ANSWER_EXEC as global option */
226 ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts);
229 ast_dial_set_global_timeout(dial, timeout * 1000);
232 if (ast_test_flag(&flags, PAGE_IGNORE_FORWARDS)) {
233 ast_dial_option_global_enable(dial, AST_DIAL_OPTION_DISABLE_CALL_FORWARDING, NULL);
236 /* Run this dial in async mode */
237 ast_dial_run(dial, chan, 1);
239 /* Put in our dialing array */
240 dial_list[pos++] = dial;
243 if (!ast_test_flag(&flags, PAGE_QUIET)) {
244 res = ast_streamfile(chan, "beep", chan->language);
246 res = ast_waitstream(chan, "");
250 snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"),
251 (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
252 pbx_exec(chan, app, meetmeopts);
255 /* Go through each dial attempt cancelling, joining, and destroying */
256 for (i = 0; i < pos; i++) {
257 struct ast_dial *dial = dial_list[i];
259 /* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */
262 /* Hangup all channels */
263 ast_dial_hangup(dial);
265 /* Destroy dialing structure */
266 ast_dial_destroy(dial);
272 static int unload_module(void)
274 return ast_unregister_application(app_page);
277 static int load_module(void)
279 return ast_register_application_xml(app_page, page_exec);
282 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");