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, 217 insertions, 0 deletions
diff --git a/1.4.23-rc4/apps/app_page.c b/1.4.23-rc4/apps/app_page.c new file mode 100644 index 000000000..c94e1b11a --- /dev/null +++ b/1.4.23-rc4/apps/app_page.c @@ -0,0 +1,217 @@ +/* + * 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"); + |