asterisk/apps/app_page.c
Matt Jordan 4a58261694 git migration: Refactor the ASTERISK_FILE_VERSION macro
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.

Specifically, it does the following:

* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
  remove passing the version in with the macro. Other facilities
  than 'core show file version' make use of the file names, such as
  setting a debug level only on a specific file. As such, the act of
  registering source files with the Asterisk core still has use. The
  macro rename now reflects the new macro purpose.

* main/asterisk:
  - Refactor the file_version structure to reflect that it no longer
    tracks a version field.
  - Remove the "core show file version" CLI command. Without the file
    version, it is no longer useful.
  - Remove the ast_file_version_find function. The file version is no
    longer tracked.
  - Rename ast_register_file_version/ast_unregister_file_version to
    ast_register_file/ast_unregister_file, respectively.

* main/manager: Remove value from the Version key of the ModuleCheck
  Action. The actual key itself has not been removed, as doing so would
  absolutely constitute a backwards incompatible change. However, since
  the file version is no longer tracked, there is no need to attempt to
  include it in the Version key.

* UPGRADE: Add notes for:
  - Modification to the ModuleCheck AMI Action
  - Removal of the "core show file version" CLI command

Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
2015-04-13 03:48:57 -04:00

431 lines
13 KiB
C

/*
* 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>app_confbridge</depend>
<support_level>core</support_level>
***/
#include "asterisk.h"
ASTERISK_REGISTER_FILE()
#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"
/*** DOCUMENTATION
<application name="Page" language="en_US">
<synopsis>
Page series of phones
</synopsis>
<syntax>
<parameter name="Technology/Resource" required="true" argsep="&amp;">
<argument name="Technology/Resource" required="true">
<para>Specification of the device(s) to dial. These must be in the format of
<literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
represents a particular channel driver, and <replaceable>Resource</replaceable> represents a resource
available to that particular channel driver.</para>
</argument>
<argument name="Technology2/Resource2" multiple="true">
<para>Optional extra devices to dial in parallel</para>
<para>If you need more than one, enter them as Technology2/Resource2&amp;
Technology3/Resourse3&amp;.....</para>
</argument>
</parameter>
<parameter name="options">
<optionlist>
<option name="b" argsep="^">
<para>Before initiating an outgoing call, Gosub to the specified
location using the newly created channel. The Gosub will be
executed for each destination channel.</para>
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" hasparams="optional" argsep="^">
<argument name="arg1" multiple="true" required="true" />
<argument name="argN" />
</argument>
</option>
<option name="B" argsep="^">
<para>Before initiating the outgoing call(s), Gosub to the specified
location using the current channel.</para>
<argument name="context" required="false" />
<argument name="exten" required="false" />
<argument name="priority" required="true" hasparams="optional" argsep="^">
<argument name="arg1" multiple="true" required="true" />
<argument name="argN" />
</argument>
</option>
<option name="d">
<para>Full duplex audio</para>
</option>
<option name="i">
<para>Ignore attempts to forward the call</para>
</option>
<option name="q">
<para>Quiet, do not play beep to caller</para>
</option>
<option name="r">
<para>Record the page into a file (<literal>CONFBRIDGE(bridge,record_conference)</literal>)</para>
</option>
<option name="s">
<para>Only dial a channel if its device state says that it is <literal>NOT_INUSE</literal></para>
</option>
<option name="A">
<argument name="x" required="true">
<para>The announcement to playback to all devices</para>
</argument>
<para>Play an announcement to all paged participants</para>
</option>
<option name="n">
<para>Do not play announcement to caller (alters <literal>A(x)</literal> behavior)</para>
</option>
</optionlist>
</parameter>
<parameter name="timeout">
<para>Specify the length of time that the system will attempt to connect a call.
After this duration, any page calls that have not been answered will be hung up by the
system.</para>
</parameter>
</syntax>
<description>
<para>Places outbound calls to the given <replaceable>technology</replaceable>/<replaceable>resource</replaceable>
and dumps them into a conference bridge as muted participants. The original
caller is dumped into the conference as a speaker and the room is
destroyed when the original caller leaves.</para>
</description>
<see-also>
<ref type="application">ConfBridge</ref>
</see-also>
</application>
***/
static const char * const app_page= "Page";
enum page_opt_flags {
PAGE_DUPLEX = (1 << 0),
PAGE_QUIET = (1 << 1),
PAGE_RECORD = (1 << 2),
PAGE_SKIP = (1 << 3),
PAGE_IGNORE_FORWARDS = (1 << 4),
PAGE_ANNOUNCE = (1 << 5),
PAGE_NOCALLERANNOUNCE = (1 << 6),
PAGE_PREDIAL_CALLEE = (1 << 7),
PAGE_PREDIAL_CALLER = (1 << 8),
};
enum {
OPT_ARG_ANNOUNCE = 0,
OPT_ARG_PREDIAL_CALLEE = 1,
OPT_ARG_PREDIAL_CALLER = 2,
OPT_ARG_ARRAY_SIZE = 3,
};
AST_APP_OPTIONS(page_opts, {
AST_APP_OPTION_ARG('b', PAGE_PREDIAL_CALLEE, OPT_ARG_PREDIAL_CALLEE),
AST_APP_OPTION_ARG('B', PAGE_PREDIAL_CALLER, OPT_ARG_PREDIAL_CALLER),
AST_APP_OPTION('d', PAGE_DUPLEX),
AST_APP_OPTION('q', PAGE_QUIET),
AST_APP_OPTION('r', PAGE_RECORD),
AST_APP_OPTION('s', PAGE_SKIP),
AST_APP_OPTION('i', PAGE_IGNORE_FORWARDS),
AST_APP_OPTION_ARG('A', PAGE_ANNOUNCE, OPT_ARG_ANNOUNCE),
AST_APP_OPTION('n', PAGE_NOCALLERANNOUNCE),
});
/* We use this structure as a way to pass this to all dialed channels */
struct page_options {
char *opts[OPT_ARG_ARRAY_SIZE];
struct ast_flags flags;
};
/*!
* \internal
* \brief Setup the page bridge profile.
*
* \param chan Setup bridge profile on this channel.
* \param options Options to setup bridge profile.
*
* \return Nothing
*/
static void setup_profile_bridge(struct ast_channel *chan, struct page_options *options)
{
/* Use default_bridge as a starting point */
ast_func_write(chan, "CONFBRIDGE(bridge,template)", "");
if (ast_test_flag(&options->flags, PAGE_RECORD)) {
ast_func_write(chan, "CONFBRIDGE(bridge,record_conference)", "yes");
}
}
/*!
* \internal
* \brief Setup the paged user profile.
*
* \param chan Setup user profile on this channel.
* \param options Options to setup paged user profile.
*
* \return Nothing
*/
static void setup_profile_paged(struct ast_channel *chan, struct page_options *options)
{
/* Use default_user as a starting point */
ast_func_write(chan, "CONFBRIDGE(user,template)", "");
ast_func_write(chan, "CONFBRIDGE(user,quiet)", "yes");
ast_func_write(chan, "CONFBRIDGE(user,end_marked)", "yes");
if (!ast_test_flag(&options->flags, PAGE_DUPLEX)) {
ast_func_write(chan, "CONFBRIDGE(user,startmuted)", "yes");
}
if (ast_test_flag(&options->flags, PAGE_ANNOUNCE)
&& !ast_strlen_zero(options->opts[OPT_ARG_ANNOUNCE])) {
ast_func_write(chan, "CONFBRIDGE(user,announcement)", options->opts[OPT_ARG_ANNOUNCE]);
}
}
/*!
* \internal
* \brief Setup the caller user profile.
*
* \param chan Setup user profile on this channel.
* \param options Options to setup caller user profile.
*
* \return Nothing
*/
static void setup_profile_caller(struct ast_channel *chan, struct page_options *options)
{
/* Use default_user as a starting point if not already setup. */
ast_func_write(chan, "CONFBRIDGE(user,template)", "");
ast_func_write(chan, "CONFBRIDGE(user,quiet)", "yes");
ast_func_write(chan, "CONFBRIDGE(user,marked)", "yes");
if (!ast_test_flag(&options->flags, PAGE_NOCALLERANNOUNCE)
&& ast_test_flag(&options->flags, PAGE_ANNOUNCE)
&& !ast_strlen_zero(options->opts[OPT_ARG_ANNOUNCE])) {
ast_func_write(chan, "CONFBRIDGE(user,announcement)", options->opts[OPT_ARG_ANNOUNCE]);
}
}
static void page_state_callback(struct ast_dial *dial)
{
struct ast_channel *chan;
struct page_options *options;
if (ast_dial_state(dial) != AST_DIAL_RESULT_ANSWERED ||
!(chan = ast_dial_answered(dial)) ||
!(options = ast_dial_get_user_data(dial))) {
return;
}
setup_profile_bridge(chan, options);
setup_profile_paged(chan, options);
}
static int page_exec(struct ast_channel *chan, const char *data)
{
char *tech, *resource, *tmp;
char confbridgeopts[128], originator[AST_CHANNEL_NAME];
struct page_options options = { { 0, }, { 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;
int timeout = 0;
char *parse;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(devices);
AST_APP_ARG(options);
AST_APP_ARG(timeout);
);
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("ConfBridge"))) {
ast_log(LOG_WARNING, "There is no ConfBridge application available!\n");
return -1;
};
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
ast_copy_string(originator, ast_channel_name(chan), sizeof(originator));
if ((tmp = strchr(originator, '-'))) {
*tmp = '\0';
}
if (!ast_strlen_zero(args.options)) {
ast_app_parse_options(page_opts, &options.flags, options.opts, args.options);
}
if (!ast_strlen_zero(args.timeout)) {
timeout = atoi(args.timeout);
}
snprintf(confbridgeopts, sizeof(confbridgeopts), "ConfBridge,%u", confid);
/* Count number of extensions in list by number of ampersands + 1 */
num_dials = 1;
tmp = args.devices;
while (*tmp) {
if (*tmp == '&') {
num_dials++;
}
tmp++;
}
if (!(dial_list = ast_calloc(num_dials, sizeof(struct ast_dial *)))) {
ast_log(LOG_ERROR, "Can't allocate %ld bytes for dial list\n", (long)(sizeof(struct ast_dial *) * num_dials));
return -1;
}
if (ast_test_flag(&options.flags, PAGE_PREDIAL_CALLER)
&& !ast_strlen_zero(options.opts[OPT_ARG_PREDIAL_CALLER])) {
ast_replace_subargument_delimiter(options.opts[OPT_ARG_PREDIAL_CALLER]);
ast_app_exec_sub(NULL, chan, options.opts[OPT_ARG_PREDIAL_CALLER], 0);
}
/* Go through parsing/calling each device */
while ((tech = strsep(&args.devices, "&"))) {
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(&options.flags, PAGE_SKIP)) {
state = ast_device_state(tech);
if (state == AST_DEVICE_UNKNOWN) {
ast_log(LOG_WARNING, "Destination '%s' has device state '%s'. Paging anyway.\n", tech, ast_devstate2str(state));
} else if (state != AST_DEVICE_NOT_INUSE) {
ast_log(LOG_WARNING, "Destination '%s' has device state '%s'.\n", tech, ast_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 */
if (ast_dial_append(dial, tech, resource, NULL) == -1) {
ast_log(LOG_ERROR, "Failed to add %s to outbound dial\n", tech);
ast_dial_destroy(dial);
continue;
}
/* Set ANSWER_EXEC as global option */
ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, confbridgeopts);
if (ast_test_flag(&options.flags, PAGE_PREDIAL_CALLEE)
&& !ast_strlen_zero(options.opts[OPT_ARG_PREDIAL_CALLEE])) {
ast_dial_option_global_enable(dial, AST_DIAL_OPTION_PREDIAL, options.opts[OPT_ARG_PREDIAL_CALLEE]);
}
if (timeout) {
ast_dial_set_global_timeout(dial, timeout * 1000);
}
if (ast_test_flag(&options.flags, PAGE_IGNORE_FORWARDS)) {
ast_dial_option_global_enable(dial, AST_DIAL_OPTION_DISABLE_CALL_FORWARDING, NULL);
}
ast_dial_set_state_callback(dial, &page_state_callback);
ast_dial_set_user_data(dial, &options);
/* 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(&options.flags, PAGE_QUIET)) {
res = ast_streamfile(chan, "beep", ast_channel_language(chan));
if (!res)
res = ast_waitstream(chan, "");
}
if (!res) {
setup_profile_bridge(chan, &options);
setup_profile_caller(chan, &options);
snprintf(confbridgeopts, sizeof(confbridgeopts), "%u", confid);
pbx_exec(chan, app, confbridgeopts);
}
/* 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 ConfBridge 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);
return -1;
}
static int unload_module(void)
{
return ast_unregister_application(app_page);
}
static int load_module(void)
{
return ast_register_application_xml(app_page, page_exec);
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Page Multiple Phones");