From f8247040e6231c4b3b5099ea3a526348b7941566 Mon Sep 17 00:00:00 2001 From: russell Date: Sat, 19 Jan 2008 00:19:29 +0000 Subject: Creating tag for the release of asterisk-1.6.0-beta1 git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.0-beta1@99163 f38db490-d61c-443f-a65b-d21fe96a405b --- trunk/apps/app_page.c | 193 ++++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 193 insertions(+) create mode 100644 trunk/apps/app_page.c (limited to 'trunk/apps/app_page.c') diff --git a/trunk/apps/app_page.c b/trunk/apps/app_page.c new file mode 100644 index 000000000..2ea367edd --- /dev/null +++ b/trunk/apps/app_page.c @@ -0,0 +1,193 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (c) 2004 - 2006 Digium, Inc. All rights reserved. + * + * Mark Spencer + * + * 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 + * + * \ingroup applications + */ + +/*** MODULEINFO + zaptel + app_meetme + ***/ + +#include "asterisk.h" + +ASTERISK_FILE_VERSION(__FILE__, "$Revision$") + +#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/devicestate.h" +#include "asterisk/dial.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" +" s - only dial channel if devicestate says it is not in use\n"; + +enum { + PAGE_DUPLEX = (1 << 0), + PAGE_QUIET = (1 << 1), + PAGE_RECORD = (1 << 2), + PAGE_SKIP = (1 << 3), +} 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), + AST_APP_OPTION('s', PAGE_SKIP), +}); + +#define MAX_DIALS 128 + +static int page_exec(struct ast_channel *chan, void *data) +{ + char *options, *tech, *resource, *tmp; + char meetmeopts[88], originator[AST_CHANNEL_NAME], *opts[0]; + struct ast_flags flags = { 0 }; + unsigned int confid = ast_random(); + struct ast_app *app; + int res = 0, pos = 0, i = 0; + struct ast_dial *dials[MAX_DIALS]; + + if (ast_strlen_zero(data)) { + ast_log(LOG_WARNING, "This application requires at least one argument (destination(s) to page)\n"); + return -1; + } + + if (!(app = pbx_findapp("MeetMe"))) { + ast_log(LOG_WARNING, "There is no MeetMe application available!\n"); + 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, opts, 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" : "") ); + + /* Go through parsing/calling each device */ + while ((tech = strsep(&tmp, "&"))) { + int state = 0; + 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; + } + + /* Ensure device is not in use if skip option is enabled */ + if (ast_test_flag(&flags, PAGE_SKIP) && (state = ast_device_state(tech)) != AST_DEVICE_NOT_INUSE) { + ast_log(LOG_WARNING, "Destination '%s' has device state '%s'.\n", tech, devstate2str(state)); + 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 */ + dials[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 = dials[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); + } + + return -1; +} + +static int unload_module(void) +{ + return ast_unregister_application(app_page); +} + +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"); + -- cgit v1.2.3