Files
asterisk/apps/app_page.c
Matthew Jordan 3edf245601 Fix a variety of memory leaks
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

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-18 13:58:23 +00:00

322 lines
9.4 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>dahdi</depend>
<depend>app_meetme</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="&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 inparallel</para>
<para>If you need more then one enter them as Technology2/Resource2&amp;
Technology3/Resourse3&amp;.....</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 (meetme 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">MeetMe</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),
});
static int page_exec(struct ast_channel *chan, const char *data)
{
char *tech, *resource, *tmp;
char meetmeopts[128], originator[AST_CHANNEL_NAME], *opts[OPT_ARG_ARRAY_SIZE];
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;
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("MeetMe"))) {
ast_log(LOG_WARNING, "There is no MeetMe application available!\n");
return -1;
};
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
ast_copy_string(originator, chan->name, sizeof(originator));
if ((tmp = strchr(originator, '-'))) {
*tmp = '\0';
}
if (!ast_strlen_zero(args.options)) {
ast_app_parse_options(page_opts, &flags, opts, args.options);
} else {
/* opts must be initialized if there wasn't an options string. */
for (i = 0; i < OPT_ARG_ARRAY_SIZE; i++) {
opts[i] = NULL;
}
}
if (!ast_strlen_zero(args.timeout)) {
timeout = atoi(args.timeout);
}
if (ast_test_flag(&flags, PAGE_ANNOUNCE) && !ast_strlen_zero(opts[OPT_ARG_ANNOUNCE])) {
snprintf(meetmeopts, sizeof(meetmeopts), "MeetMe,%ud,%s%sqxdw(5)G(%s)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"),
(ast_test_flag(&flags, PAGE_RECORD) ? "r" : ""), opts[OPT_ARG_ANNOUNCE] );
} else {
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;
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(&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, meetmeopts);
if (timeout) {
ast_dial_set_global_timeout(dial, timeout * 1000);
}
if (ast_test_flag(&flags, PAGE_IGNORE_FORWARDS)) {
ast_dial_option_global_enable(dial, AST_DIAL_OPTION_DISABLE_CALL_FORWARDING, NULL);
}
/* 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) {
/* Default behaviour */
snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqxd", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"),
(ast_test_flag(&flags, PAGE_RECORD) ? "r" : "") );
if (ast_test_flag(&flags, PAGE_ANNOUNCE) && !ast_strlen_zero(opts[OPT_ARG_ANNOUNCE]) &&
!ast_test_flag(&flags, PAGE_NOCALLERANNOUNCE)) {
snprintf(meetmeopts, sizeof(meetmeopts), "%ud,A%s%sqxdG(%s)", confid, (ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "t"),
(ast_test_flag(&flags, PAGE_RECORD) ? "r" : ""), opts[OPT_ARG_ANNOUNCE] );
}
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);
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");