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https://github.com/asterisk/asterisk.git
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Fix "bug" of not being able to page multiple phones
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6753 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -28,7 +28,7 @@ APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_mp3.so\
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app_test.so app_forkcdr.so app_math.so app_realtime.so \
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app_test.so app_forkcdr.so app_math.so app_realtime.so \
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app_dumpchan.so app_waitforsilence.so app_while.so app_setrdnis.so \
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app_dumpchan.so app_waitforsilence.so app_while.so app_setrdnis.so \
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app_md5.so app_readfile.so app_chanspy.so app_settransfercapability.so \
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app_md5.so app_readfile.so app_chanspy.so app_settransfercapability.so \
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app_dictate.so app_externalivr.so app_directed_pickup.so
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app_dictate.so app_externalivr.so app_directed_pickup.so app_page.so
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#
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#
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# Obsolete things...
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# Obsolete things...
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125
apps/app_page.c
Executable file
125
apps/app_page.c
Executable file
@@ -0,0 +1,125 @@
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (c) 2004 - 2005 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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/*
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*
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* Page application
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*
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <string.h>
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/options.h"
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#include "asterisk/logger.h"
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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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static char *tdesc = "Page Multiple Phones";
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static char *app_page= "Page";
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static char *page_synopsis = "Pages phones";
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static char *page_descrip =
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"Page(Technology/Resource&Technology2/Resource2)\n"
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" Places outbound calls to the given technology / resource and dumps\n"
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"them into a conference bridge as muted participants. The original\n"
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"caller is dumped into the conference as a speaker and the room is\n"
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"destroyed when the original caller leaves. Always returns -1.\n";
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STANDARD_LOCAL_USER;
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LOCAL_USER_DECL;
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static int page_exec(struct ast_channel *chan, void *data)
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{
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char *options;
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char *tech, *resource;
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char meetmeopts[80];
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unsigned int confid = rand();
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struct ast_app *app;
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if (data) {
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options = ast_strdupa((char *)data);
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if (options) {
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char *tmp = strsep(&options, "|,");
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if (options) {
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/* XXX Parse options if we had any XXX */
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}
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snprintf(meetmeopts, sizeof(meetmeopts), "%ud|mqxdw", confid);
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while(tmp && !ast_strlen_zero(tmp)) {
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tech = strsep(&tmp, "&");
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if (tech) {
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resource = strchr(tech, '/');
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if (resource) {
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*resource = '\0';
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resource++;
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ast_pbx_outgoing_app(tech, AST_FORMAT_SLINEAR, resource, 30000, "MeetMe", meetmeopts, NULL, 0, chan->cid.cid_num, chan->cid.cid_name, NULL, NULL);
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}
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}
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}
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snprintf(meetmeopts, sizeof(meetmeopts), "%ud|Atqxd", confid);
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app = pbx_findapp("Meetme");
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if (app) {
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pbx_exec(chan, app, meetmeopts, 1);
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} else
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ast_log(LOG_WARNING, "Whoa, meetme doesn't exist!\n");
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} else {
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ast_log(LOG_ERROR, "Out of memory\n");
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}
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}
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return -1;
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}
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int unload_module(void)
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{
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STANDARD_HANGUP_LOCALUSERS;
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return ast_unregister_application(app_page);
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}
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int load_module(void)
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{
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return ast_register_application(app_page, page_exec, page_synopsis, page_descrip);
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}
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char *description(void)
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{
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return tdesc;
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}
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int usecount(void)
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{
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int res;
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STANDARD_USECOUNT(res);
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return res;
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}
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char *key()
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{
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return ASTERISK_GPL_KEY;
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}
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