Merged revisions 305253 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.6.2

................
  r305253 | qwell | 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) | 17 lines
  
  Merged revisions 305252 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | 10 lines
    
    Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/))
    
    chan_iax2 and other channel drivers already had code to prevent this.  The
    attempt that app_dial was making to prevent it was not correct, so I fixed that.
    
    (closes issue #18371)
    Reported by: gbour
    Patches: 
          18371.patch uploaded by gbour (license 1162)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305254 65c4cc65-6c06-0410-ace0-fbb531ad65f3

apps/app_dial.c

@@ -1944,7 +1944,7 @@ static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast
 		struct ast_dialed_interface *di;
 		AST_LIST_HEAD(, ast_dialed_interface) *dialed_interfaces;
 		num_dialed++;
-		if (!number) {
+		if (ast_strlen_zero(number)) {
 			ast_log(LOG_WARNING, "Dial argument takes format (technology/[device:]number1)\n");
 			goto out;
 		}

channels/chan_sip.c

@@ -25142,6 +25142,12 @@ static struct ast_channel *sip_request_call(const char *type, format_t format, c
 	}
 	ast_debug(1, "Asked to create a SIP channel with formats: %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), oldformat));
 
+	if (ast_strlen_zero(dest)) {
+		ast_log(LOG_ERROR, "Unable to create channel with empty destination.\n");
+		*cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
+		return NULL;
+	}
+
 	if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE, NULL))) {
 		ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", dest);
 		*cause = AST_CAUSE_SWITCH_CONGESTION;