This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 ........ Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366881 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
350 lines
9.9 KiB
C
350 lines
9.9 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_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"
|
|
|
|
/*** DOCUMENTATION
|
|
<application name="Page" language="en_US">
|
|
<synopsis>
|
|
Page series of phones
|
|
</synopsis>
|
|
<syntax>
|
|
<parameter name="Technology/Resource" required="true" argsep="&">
|
|
<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 inparallel</para>
|
|
<para>If you need more then one enter them as Technology2/Resource2&
|
|
Technology3/Resourse3&.....</para>
|
|
</argument>
|
|
</parameter>
|
|
<parameter name="options">
|
|
<optionlist>
|
|
<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 (ConfBridge option <literal>r</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 in all devices</para>
|
|
</argument>
|
|
<para>Play an announcement simultaneously to all paged participants</para>
|
|
</option>
|
|
<option name="n">
|
|
<para>Do not play simultaneous announcement to caller (implies <literal>A(x)</literal>)</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 intercom 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 callers 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),
|
|
};
|
|
|
|
enum {
|
|
OPT_ARG_ANNOUNCE = 0,
|
|
OPT_ARG_ARRAY_SIZE = 1,
|
|
};
|
|
|
|
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),
|
|
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;
|
|
};
|
|
|
|
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;
|
|
}
|
|
|
|
ast_func_write(chan, "CONFBRIDGE(bridge,template)", "default_bridge");
|
|
|
|
if (ast_test_flag(&options->flags, PAGE_RECORD)) {
|
|
ast_func_write(chan, "CONFBRIDGE(bridge,record_conference)", "yes");
|
|
}
|
|
|
|
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]);
|
|
}
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
/* 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) == -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 (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) {
|
|
ast_func_write(chan, "CONFBRIDGE(bridge,template)", "default_bridge");
|
|
|
|
if (ast_test_flag(&options.flags, PAGE_RECORD)) {
|
|
ast_func_write(chan, "CONFBRIDGE(bridge,record_conference)", "yes");
|
|
}
|
|
|
|
ast_func_write(chan, "CONFBRIDGE(user,quiet)", "yes");
|
|
ast_func_write(chan, "CONFBRIDGE(user,marked)", "yes");
|
|
|
|
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");
|
|
|