Merged revisions 373466,373468 via svnmerge from

file:///srv/subversion/repos/asterisk/branches/10

................
  r373466 | rmudgett | 2012-09-24 15:44:27 -0500 (Mon, 24 Sep 2012) | 33 lines
  
  Fix potential reentrancy problems in chan_sip.
  
  Asterisk v1.8 and later was not as vulnerable to this issue.
  
  * Made find_call() lock each private as it processes the found dialogs.
  (Primary cause of ABE-2876)
  
  * Made the other functions that traverse the dialogs container lock each
  private as it examines them.
  
  * Fix race condition in sip_call() if the thread that sent the INVITE is
  held up long enough for a response to be processed.  The p->initid for the
  INVITE retransmission could be added after it was canceled by the response
  processing.
  
  * Made __sip_destroy() clean up resource pointers after freeing.  This is
  primarily defensive in case someone has a stale private pointer.
  
  * Removed redundant memset() in reqprep().  The call to init_req() already
  does the memset() and is the first reference to req in reqprep().
  
  * Removed useless set of req.method in transmit_invite().  The calls to
  initreqprep() and reqprep() have to do this because they memset() the req.
  
  JIRA ABE-2876
  
  ..........
  
  Merged -r373423 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
  ........
  
  Merged revisions 373424 from http://svn.asterisk.org/svn/asterisk/branches/1.8
................
  r373468 | jrose | 2012-09-24 16:05:44 -0500 (Mon, 24 Sep 2012) | 13 lines
  
  func_audiohookinherit: Document some missed sources.
  
  This patch also mentions that AUDIOHOOK_INHERIT can be used to
  transfer MixMonitor audiohooks. There is also wiki that addresses
  audiohooks and the use of AUDIOHOOK_INHERIT at the following link:
  https://wiki.asterisk.org/wiki/display/AST/Audiohooks
  
  (closes issue ASTERISK-18220)
  Reported by: Ishfaq Malik
  ........
  
  Merged revisions 373467 from http://svn.asterisk.org/svn/asterisk/branches/1.8
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@373498 65c4cc65-6c06-0410-ace0-fbb531ad65f3

apps/app_mixmonitor.c

@@ -128,6 +128,11 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 			<para>Records the audio on the current channel to the specified file.</para>
 			<para>This application does not automatically answer and should be preceeded by
 			an application such as Answer or Progress().</para>
+			<note><para>MixMonitor runs as an audiohook. In order to keep it running through
+			a transfer, AUDIOHOOK_INHERIT must be set for the channel which ran mixmonitor.
+			For more information, including dialplan configuration set for using
+			AUDIOHOOK_INHERIT with MixMonitor, see the function documentation for
+			AUDIOHOOK_INHERIT.</para></note>
 			<variablelist>
 				<variable name="MIXMONITOR_FILENAME">
 					<para>Will contain the filename used to record.</para>
@@ -139,6 +144,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 			<ref type="application">StopMixMonitor</ref>
 			<ref type="application">PauseMonitor</ref>
 			<ref type="application">UnpauseMonitor</ref>
+			<ref type="function">AUDIOHOOK_INHERIT</ref>
 		</see-also>
 	</application>
 	<application name="StopMixMonitor" language="en_US">

channels/chan_sip.c

@@ -5806,9 +5806,10 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
 		}
 
 		xmitres = transmit_invite(p, SIP_INVITE, 1, 2, uri);
-		sip_pvt_unlock(p);
-		if (xmitres == XMIT_ERROR)
+		if (xmitres == XMIT_ERROR) {
+			sip_pvt_unlock(p);
 			return -1;
+		}
 		p->invitestate = INV_CALLING;
 
 		/* Initialize auto-congest time */
@@ -5816,6 +5817,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
 								dialog_unref(_data, "dialog ptr dec when SCHED_REPLACE del op succeeded"),
 								dialog_unref(p, "dialog ptr dec when SCHED_REPLACE add failed"),
 								dialog_ref(p, "dialog ptr inc when SCHED_REPLACE add succeeded") );
+		sip_pvt_unlock(p);
 	}
 	return res;
 }
@@ -5909,6 +5911,7 @@ void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist)
 	
 	if (p->mwi) {
 		p->mwi->call = NULL;
+		p->mwi = NULL;
 	}
 
 	if (dumphistory)
@@ -5919,29 +5922,37 @@ void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist)
 			ao2_ref(p->options->outboundproxy, -1);
 		}
 		ast_free(p->options);
+		p->options = NULL;
 	}
 
 	if (p->notify) {
 		ast_variables_destroy(p->notify->headers);
 		ast_free(p->notify->content);
 		ast_free(p->notify);
+		p->notify = NULL;
 	}
 	if (p->rtp) {
 		ast_rtp_instance_destroy(p->rtp);
+		p->rtp = NULL;
 	}
 	if (p->vrtp) {
 		ast_rtp_instance_destroy(p->vrtp);
+		p->vrtp = NULL;
 	}
 	if (p->trtp) {
 		ast_rtp_instance_destroy(p->trtp);
+		p->trtp = NULL;
 	}
-	if (p->udptl)
+	if (p->udptl) {
 		ast_udptl_destroy(p->udptl);
+		p->udptl = NULL;
+	}
 	if (p->refer) {
 		if (p->refer->refer_call) {
 			p->refer->refer_call = dialog_unref(p->refer->refer_call, "unref dialog p->refer->refer_call");
 		}
 		ast_free(p->refer);
+		p->refer = NULL;
 	}
 	if (p->route) {
 		free_old_route(p->route);
@@ -5994,6 +6005,7 @@ void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist)
 	ast_string_field_free_memory(p);
 
 	ast_cc_config_params_destroy(p->cc_params);
+	p->cc_params = NULL;
 
 	if (p->epa_entry) {
 		ao2_ref(p->epa_entry, -1);
@@ -8277,10 +8289,25 @@ static enum match_req_res match_req_to_dialog(struct sip_pvt *sip_pvt_ptr, struc
 static void forked_invite_init(struct sip_request *req, const char *new_theirtag, struct sip_pvt *original, struct ast_sockaddr *addr)
 {
 	struct sip_pvt *p;
+	const char *callid;
+
+	sip_pvt_lock(original);
+	callid = ast_strdupa(original->callid);
+	sip_pvt_unlock(original);
 
-	if (!(p = sip_alloc(original->callid, addr, 1, SIP_INVITE, req)))  {
+	if (!(p = sip_alloc(callid, addr, 1, SIP_INVITE, req)))  {
 		return; /* alloc error */
 	}
+
+	/* Lock p and original private structures. */
+	sip_pvt_lock(p);
+	while (sip_pvt_trylock(original)) {
+		/* Can't use DEADLOCK_AVOIDANCE since p is an ao2 object */
+		sip_pvt_unlock(p);
+		sched_yield();
+		sip_pvt_lock(p);
+	}
+
 	p->invitestate = INV_TERMINATED;
 	p->ocseq = original->ocseq;
 	p->branch = original->branch;
@@ -8292,6 +8319,9 @@ static void forked_invite_init(struct sip_request *req, const char *new_theirtag
 	ast_string_field_set(p, uri, original->uri);
 	ast_string_field_set(p, our_contact, original->our_contact);
 	ast_string_field_set(p, fullcontact, original->fullcontact);
+
+	sip_pvt_unlock(original);
+
 	parse_ok_contact(p, req);
 	build_route(p, req, 1, 0);
 
@@ -8299,6 +8329,7 @@ static void forked_invite_init(struct sip_request *req, const char *new_theirtag
 	transmit_request(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
 
 	pvt_set_needdestroy(p, "forked request"); /* this dialog will terminate once the BYE is responed to or times out. */
+	sip_pvt_unlock(p);
 	dialog_unref(p, "setup forked invite termination");
 }
 
@@ -8478,7 +8509,9 @@ static struct sip_pvt *find_call(struct sip_request *req, struct ast_sockaddr *a
 
 		/* Iterate a list of dialogs already matched by Call-id */
 		while (iterator && (sip_pvt_ptr = ao2_iterator_next(iterator))) {
+			sip_pvt_lock(sip_pvt_ptr);
 			found = match_req_to_dialog(sip_pvt_ptr, &args);
+			sip_pvt_unlock(sip_pvt_ptr);
 
 			switch (found) {
 			case SIP_REQ_MATCH:
@@ -8502,6 +8535,7 @@ static struct sip_pvt *find_call(struct sip_request *req, struct ast_sockaddr *a
 			case SIP_REQ_NOT_MATCH:
 			default:
 				dialog_unref(sip_pvt_ptr, "pvt did not match incoming SIP msg, unref from search");
+				break;
 			}
 		}
 		if (iterator) {
@@ -10765,8 +10799,6 @@ static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, ui
 	int is_strict = FALSE;		/*!< Strict routing flag */
 	int is_outbound = ast_test_flag(&p->flags[0], SIP_OUTGOING);	/* Session direction */
 
-	memset(req, 0, sizeof(struct sip_request));
-
 	snprintf(p->lastmsg, sizeof(p->lastmsg), "Tx: %s", sip_methods[sipmethod].text);
 
 	if (!seqno) {
@@ -12269,19 +12301,21 @@ static void copy_request(struct sip_request *dst, const struct sip_request *src)
 
 	/* copy the entire request then restore the original data and content
 	 * members from the dst request */
-	memcpy(dst, src, sizeof(*dst));
+	*dst = *src;
 	dst->data = duplicate;
 	dst->content = duplicate_content;
 
 	/* copy the data into the dst request */
-	if (!dst->data && !(dst->data = ast_str_create(ast_str_strlen(src->data) + 1)))
+	if (!dst->data && !(dst->data = ast_str_create(ast_str_strlen(src->data) + 1))) {
 		return;
+	}
 	ast_str_copy_string(&dst->data, src->data);
 
 	/* copy the content into the dst request (if it exists) */
 	if (src->content) {
-		if (!dst->content && !(dst->content = ast_str_create(ast_str_strlen(src->content) + 1)))
+		if (!dst->content && !(dst->content = ast_str_create(ast_str_strlen(src->content) + 1))) {
 			return;
+		}
 		ast_str_copy_string(&dst->content, src->content);
 	}
 }
@@ -12754,8 +12788,7 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init,
 {
 	struct sip_request req;
 	struct ast_variable *var;
-	
-	req.method = sipmethod;
+
 	if (init) {/* Bump branch even on initial requests */
 		p->branch ^= ast_random();
 		p->invite_branch = p->branch;
@@ -18825,13 +18858,18 @@ static int show_chanstats_cb(void *__cur, void *__arg, int flags)
 	char durbuf[10];
 	int duration;
 	int durh, durm, durs;
-	struct ast_channel *c = cur->owner;
+	struct ast_channel *c;
 	struct __show_chan_arg *arg = __arg;
 	int fd = arg->fd;
 
+	sip_pvt_lock(cur);
+	c = cur->owner;
 
-	if (cur->subscribed != NONE) /* Subscriptions */
+	if (cur->subscribed != NONE) {
+		/* Subscriptions */
+		sip_pvt_unlock(cur);
 		return 0;	/* don't care, we scan all channels */
+	}
 
 	if (!cur->rtp) {
 		if (sipdebug) {
@@ -18840,10 +18878,12 @@ static int show_chanstats_cb(void *__cur, void *__arg, int flags)
 				invitestate2string[cur->invitestate].desc,
 				"-- No RTP active");
 		}
+		sip_pvt_unlock(cur);
 		return 0;	/* don't care, we scan all channels */
 	}
 
 	if (ast_rtp_instance_get_stats(cur->rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
+		sip_pvt_unlock(cur);
 		ast_log(LOG_WARNING, "Could not get RTP stats.\n");
 		return 0;
 	}
@@ -18874,6 +18914,7 @@ static int show_chanstats_cb(void *__cur, void *__arg, int flags)
 		stats.txjitter
 	);
 	arg->numchans++;
+	sip_pvt_unlock(cur);
 
 	return 0;	/* don't care, we scan all channels */
 }
@@ -19211,8 +19252,11 @@ static int show_channels_cb(void *__cur, void *__arg, int flags)
 {
 	struct sip_pvt *cur = __cur;
 	struct __show_chan_arg *arg = __arg;
-	const struct ast_sockaddr *dst = sip_real_dst(cur);
-	
+	const struct ast_sockaddr *dst;
+
+	sip_pvt_lock(cur);
+	dst = sip_real_dst(cur);
+
 	/* XXX indentation preserved to reduce diff. Will be fixed later */
 	if (cur->subscribed == NONE && !arg->subscriptions) {
 		/* set if SIP transfer in progress */
@@ -19247,6 +19291,7 @@ static int show_channels_cb(void *__cur, void *__arg, int flags)
 			);
 		arg->numchans++;
 	}
+	sip_pvt_unlock(cur);
 	return 0;	/* don't care, we scan all channels */
 }
 

funcs/func_audiohookinherit.c

@@ -53,7 +53,9 @@
 					<enum name="Chanspy" />
 					<enum name="Volume" />
 					<enum name="Speex" />
+					<enum name="pitch_shift" />
 					<enum name="JACK_HOOK" />
+					<enum name="Mute" />
 				</enumlist>
 				<para>Note that the names are not case-sensitive</para>
 			</parameter>