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
........

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

1 files changed, 2 insertions, 0 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5d0cf0bdd..a21ebf394 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -7657,6 +7657,7 @@ static struct ast_frame *sip_read(struct ast_channel *ast)
 				if (ast_async_goto(ast, target_context, "fax", 1)) {
 					ast_log(LOG_NOTICE, "Failed to async goto '%s' into fax of '%s'\n", ast_channel_name(ast), target_context);
 				}
+				ast_frfree(fr);
 				fr = &ast_null_frame;
 			} else {
 				ast_channel_lock(ast);
@@ -7668,6 +7669,7 @@ static struct ast_frame *sip_read(struct ast_channel *ast)
 
 	/* Only allow audio through if they sent progress with SDP, or if the channel is actually answered */
 	if (fr && fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast_channel_state(ast) != AST_STATE_UP) {
+		ast_frfree(fr);
 		fr = &ast_null_frame;
 	}