diff options
Diffstat (limited to '1.4.23-rc4/apps/app_page.c')
| -rw-r--r-- | 1.4.23-rc4/apps/app_page.c | 217 |
1 files changed, 0 insertions, 217 deletions
diff --git a/1.4.23-rc4/apps/app_page.c b/1.4.23-rc4/apps/app_page.c deleted file mode 100644 index c94e1b11a..000000000 --- a/1.4.23-rc4/apps/app_page.c +++ /dev/null @@ -1,217 +0,0 @@ -/* - * Asterisk -- An open source telephony toolkit. - * - * Copyright (c) 2004 - 2006 Digium, Inc. All rights reserved. - * - * Mark Spencer <markster@digium.com> - * - * This code is released under the GNU General Public License - * version 2.0. See LICENSE for more information. - * - * 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. - * - */ - -/*! \file - * - * \brief page() - Paging application - * - * \author Mark Spencer <markster@digium.com> - * - * \ingroup applications - */ - -/*** MODULEINFO - <depend>dahdi</depend> - <depend>app_meetme</depend> - ***/ - -#include "asterisk.h" - -ASTERISK_FILE_VERSION(__FILE__, "$Revision$") - -#include <stdio.h> -#include <stdlib.h> -#include <unistd.h> -#include <string.h> -#include <errno.h> - -#include "asterisk/options.h" -#include "asterisk/logger.h" -#include "asterisk/channel.h" -#include "asterisk/pbx.h" -#include "asterisk/module.h" -#include "asterisk/file.h" -#include "asterisk/app.h" -#include "asterisk/chanvars.h" -#include "asterisk/utils.h" -#include "asterisk/dial.h" -#include "asterisk/devicestate.h" - -static const char *app_page= "Page"; - -static const char *page_synopsis = "Pages phones"; - -static const char *page_descrip = -"Page(Technology/Resource&Technology2/Resource2[|options])\n" -" Places outbound calls to the given technology / resource and dumps\n" -"them into a conference bridge as muted participants. The original\n" -"caller is dumped into the conference as a speaker and the room is\n" -"destroyed when the original caller leaves. Valid options are:\n" -" d - full duplex audio\n" -" q - quiet, do not play beep to caller\n" -" r - record the page into a file (see 'r' for app_meetme)\n"; - -enum { - PAGE_DUPLEX = (1 << 0), - PAGE_QUIET = (1 << 1), - PAGE_RECORD = (1 << 2), -} page_opt_flags; - -AST_APP_OPTIONS(page_opts, { - AST_APP_OPTION('d', PAGE_DUPLEX), - AST_APP_OPTION('q', PAGE_QUIET), - AST_APP_OPTION('r', PAGE_RECORD), -}); - - -static int page_exec(struct ast_channel *chan, void *data) -{ - struct ast_module_user *u; - char *options, *tech, *resource, *tmp, *tmp2; - char meetmeopts[88], originator[AST_CHANNEL_NAME]; - struct ast_flags flags = { 0 }; - unsigned int confid = ast_random(); - struct ast_app *app; - int res = 0, pos = 0, i = 0; - struct ast_dial **dial_list; - unsigned int num_dials; - - if (ast_strlen_zero(data)) { - ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n"); - return -1; - } - - u = ast_module_user_add(chan); - - if (!(app = pbx_findapp("MeetMe"))) { - ast_log(LOG_WARNING, "There is no MeetMe application available!\n"); - ast_module_user_remove(u); - return -1; - }; - - options = ast_strdupa(data); - - ast_copy_string(originator, chan->name, sizeof(originator)); - if ((tmp = strchr(originator, '-'))) - *tmp = '\0'; - - tmp = strsep(&options, "|"); - if (options) - ast_app_parse_options(page_opts, &flags, NULL, options); - - snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe|%ud|%s%sqxdw(5)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"), - (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") ); - - /* Count number of extensions in list by number of ampersands + 1 */ - num_dials = 1; - tmp2 = tmp; - while (*tmp2 && *tmp2++ == '&') { - num_dials++; - } - - if (!(dial_list = ast_calloc(num_dials, sizeof(void *)))) { - ast_log(LOG_ERROR, "Can't allocate %ld bytes for dial list\n", (long)(sizeof(void *) * num_dials)); - ast_module_user_remove(u); - return -1; - } - - /* Go through parsing/calling each device */ - while ((tech = strsep(&tmp, "&"))) { - struct ast_dial *dial = NULL; - - /* don't call the originating device */ - if (!strcasecmp(tech, originator)) - continue; - - /* If no resource is available, continue on */ - if (!(resource = strchr(tech, '/'))) { - ast_log(LOG_WARNING, "Incomplete destination '%s' supplied.\n", tech); - continue; - } - - *resource++ = '\0'; - - /* Create a dialing structure */ - if (!(dial = ast_dial_create())) { - ast_log(LOG_WARNING, "Failed to create dialing structure.\n"); - continue; - } - - /* Append technology and resource */ - ast_dial_append(dial, tech, resource); - - /* Set ANSWER_EXEC as global option */ - ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, meetmeopts); - - /* Run this dial in async mode */ - ast_dial_run(dial, chan, 1); - - /* Put in our dialing array */ - dial_list[pos++] = dial; - } - - if (!ast_test_flag(&flags, PAGE_QUIET)) { - res = ast_streamfile(chan, "beep", chan->language); - if (!res) - res = ast_waitstream(chan, ""); - } - - if (!res) { - snprintf(meetmeopts, sizeof(meetmeopts), "%ud|A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"), - (ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") ); - pbx_exec(chan, app, meetmeopts); - } - - /* Go through each dial attempt cancelling, joining, and destroying */ - for (i = 0; i < pos; i++) { - struct ast_dial *dial = dial_list[i]; - - /* We have to wait for the async thread to exit as it's possible Meetme won't throw them out immediately */ - ast_dial_join(dial); - - /* Hangup all channels */ - ast_dial_hangup(dial); - - /* Destroy dialing structure */ - ast_dial_destroy(dial); - } - - ast_free(dial_list); - ast_module_user_remove(u); - - return -1; -} - -static int unload_module(void) -{ - int res; - - res = ast_unregister_application(app_page); - - ast_module_user_hangup_all(); - - return res; -} - -static int load_module(void) -{ - return ast_register_application(app_page, page_exec, page_synopsis, page_descrip); -} - -AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones"); - |