Code cleanup

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@2919 65c4cc65-6c06-0410-ace0-fbb531ad65f3

app.c

@@ -260,7 +260,8 @@ int ast_app_messagecount(const char *mailbox, int *newmsgs, int *oldmsgs)
 	return 0;
 }
 
-int ast_dtmf_stream(struct ast_channel *chan,struct ast_channel *peer,char *digits,int between) {
+int ast_dtmf_stream(struct ast_channel *chan,struct ast_channel *peer,char *digits,int between) 
+{
 	char *ptr=NULL;
 	int res=0;
 	struct ast_frame f;
@@ -286,8 +287,7 @@ int ast_dtmf_stream(struct ast_channel *chan,struct ast_channel *peer,char *digi
 				f.src = "ast_dtmf_stream";
 				if (strchr("0123456789*#abcdABCD",*ptr)==NULL) {
 					ast_log(LOG_WARNING, "Illegal DTMF character '%c' in string. (0-9*#aAbBcCdD allowed)\n",*ptr);
-	} 
-	else {
+				} else {
 					res = ast_write(chan, &f);
 					if (res) 
 						break;
@@ -301,6 +301,5 @@ int ast_dtmf_stream(struct ast_channel *chan,struct ast_channel *peer,char *digi
 		if (peer)
 			res = ast_autoservice_stop(peer);
 	}
-
 	return res;
 }

apps/app_dial.c

@@ -25,6 +25,7 @@
 #include <asterisk/musiconhold.h>
 #include <asterisk/callerid.h>
 #include <asterisk/utils.h>
+#include <asterisk/app.h>
 #include <stdlib.h>
 #include <errno.h>
 #include <unistd.h>
@@ -477,10 +478,9 @@ static int dial_exec(struct ast_channel *chan, void *data)
 			sdtmfptr = strchr(sdtmfdata, ')');
 			if (sdtmfptr)
 				*sdtmfptr = '\0';
-		  else {
+			else 
 				ast_log(LOG_WARNING, "D( Data lacking trailing ')'\n");
 		}
-		}
 		
 		/* XXX LIMIT SUPPORT */
 		if ((limitptr = strstr(transfer, "L("))) {
@@ -494,9 +494,8 @@ static int dial_exec(struct ast_channel *chan, void *data)
 			limitptr = strchr(limitdata, ')');
 			if (limitptr)
 				*limitptr = '\0';
-		  else {
+			else
 				ast_log(LOG_WARNING, "Limit Data lacking trailing ')'\n");
-		  }
 
 			var = pbx_builtin_getvar_helper(chan,"LIMIT_PLAYAUDIO_CALLER");
 			play_to_caller = var ? ast_true(var) : 1;
@@ -544,8 +543,7 @@ static int dial_exec(struct ast_channel *chan, void *data)
 			if (!play_warning && !start_sound && !end_sound && timelimit) { 
 				calldurationlimit=timelimit/1000;
 				timelimit=play_to_caller=play_to_callee=play_warning=warning_freq=0;
-		  }
-		  else if (option_verbose > 2) {
+			} else if (option_verbose > 2) {
 				ast_verbose(VERBOSE_PREFIX_3"Limit Data:\n");
 				ast_verbose(VERBOSE_PREFIX_3"timelimit=%ld\n",timelimit);
 				ast_verbose(VERBOSE_PREFIX_3"play_warning=%ld\n",play_warning);
@@ -556,7 +554,6 @@ static int dial_exec(struct ast_channel *chan, void *data)
 				ast_verbose(VERBOSE_PREFIX_3"warning_sound=%s\n",warning_sound ? warning_sound : "UNDEF");
 				ast_verbose(VERBOSE_PREFIX_3"end_sound=%s\n",end_sound ? end_sound : "UNDEF");
 			}
-		  
 		}
 		
 		/* XXX ANNOUNCE SUPPORT */
@@ -845,8 +842,7 @@ static int dial_exec(struct ast_channel *chan, void *data)
 			
 			// Ok, done. stop autoservice
 			res = ast_autoservice_stop(chan);
-		}
-		else 
+		} else
 			res = 0;
 
 		if (!res) {
@@ -854,8 +850,7 @@ static int dial_exec(struct ast_channel *chan, void *data)
 				time(&now);
 				chan->whentohangup = now + calldurationlimit;
 			}
-
-		  if(strlen(sdtmfdata)) 
+			if (!ast_strlen_zero(sdtmfdata)) 
 				res = ast_dtmf_stream(peer,chan,sdtmfdata,0);
 		}
 		
@@ -873,8 +868,7 @@ static int dial_exec(struct ast_channel *chan, void *data)
 			config.end_sound = end_sound;
 			config.start_sound = start_sound;
 			res = ast_bridge_call(chan,peer,&config);
-		}
-		else 
+		} else 
 			res = -1;
 		
 		if (res != AST_PBX_NO_HANGUP_PEER)

include/asterisk/app.h

@@ -48,7 +48,7 @@ extern int ast_app_messagecount(const char *mailbox, int *newmsgs, int *oldmsgs)
 //! Safely spawn an external program while closingn file descriptors
 extern int ast_safe_system(const char *s);
 
-// send DTMF to chan (optionally entertain peer)  
+//! Send DTMF to chan (optionally entertain peer)  
 int ast_dtmf_stream(struct ast_channel *chan, struct ast_channel *peer, char *digits, int between);
 
 #if defined(__cplusplus) || defined(c_plusplus)