Version 0.1.2 from FTP

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149 65c4cc65-6c06-0410-ace0-fbb531ad65f3

apps/app_intercom.c

@@ -59,7 +59,7 @@ static int write_audio(short *data, int len)
 		pthread_mutex_unlock(&sound_lock);
         return -1;
     }
-		res = write(sound, data, len);
+	res = write(sound, data, len);
 	pthread_mutex_unlock(&sound_lock);
 	return res;
 }
@@ -97,7 +97,7 @@ static int create_audio()
 		return -1;
 	}
 	if (fmt != desired) {
-		ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n");
+		ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
 	}
 #if 1
 	/* 2 bytes * 15 units of 2^5 = 32 bytes per buffer */

include/asterisk/pbx.h

@@ -56,6 +56,11 @@ int ast_unregister_application(char *app);
 /* If an extension exists, return non-zero */
 int ast_exists_extension(struct ast_channel *c, char *context, char *exten, int priority);
 
+/* If "exten" *could be* a valid extension in this context with or without
+   some more digits, return non-zero.  Basically, when this returns 0, no matter
+   what you add to exten, it's not going to be a valid extension anymore */
+int ast_canmatch_extension(struct ast_channel *c, char *context, char *exten, int priority);
+
 /* Launch a new extension (i.e. new stack) */
 int ast_spawn_extension(struct ast_channel *c, char *context, char *exten, int priority);