[asterisk-commits] qwell: trunk r305255 - in /trunk: ./ apps/app_dial.c channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: qwell
Date: Mon Jan 31 17:08:38 2011
New Revision: 305255

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=305255
Log:
Merged revisions 305254 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r305254 | qwell | 2011-01-31 17:07:00 -0600 (Mon, 31 Jan 2011) | 24 lines
  
  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)
    ........
  ................
................

Modified:
    trunk/   (props changed)
    trunk/apps/app_dial.c
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.

Modified: trunk/apps/app_dial.c
URL: http://svnview.digium.com/svn/asterisk/trunk/apps/app_dial.c?view=diff&rev=305255&r1=305254&r2=305255
==============================================================================
--- trunk/apps/app_dial.c (original)
+++ trunk/apps/app_dial.c Mon Jan 31 17:08:38 2011
@@ -1944,7 +1944,7 @@
 		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;
 		}

Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=305255&r1=305254&r2=305255
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Jan 31 17:08:38 2011
@@ -25347,6 +25347,12 @@
 	}
 	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;