[asterisk-commits] qwell: branch 1.8 r305254 - in /branches/1.8: ./ apps/ channels/

[SVN commits to the Asterisk project]

Author: qwell
Date: Mon Jan 31 17:07:00 2011
New Revision: 305254

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=305254
Log:
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:
    branches/1.8/   (props changed)
    branches/1.8/apps/app_dial.c
    branches/1.8/channels/chan_sip.c

Propchange: branches/1.8/
------------------------------------------------------------------------------
Binary property 'branch-1.6.2-merged' - no diff available.

Modified: branches/1.8/apps/app_dial.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/apps/app_dial.c?view=diff&rev=305254&r1=305253&r2=305254
==============================================================================
--- branches/1.8/apps/app_dial.c (original)
+++ branches/1.8/apps/app_dial.c Mon Jan 31 17:07:00 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: branches/1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?view=diff&rev=305254&r1=305253&r2=305254
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Mon Jan 31 17:07:00 2011
@@ -25142,6 +25142,12 @@
 	}
 	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;