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Merged revisions 305254 via svnmerge from 

https://origsvn.digium.com/svn/asterisk/branches/1.8

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  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
  
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    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)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@305255 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
qwell 2011-01-31 23:08:38 +00:00
parent e1cff4ff5a
commit 539d706d05
2 changed files with 7 additions and 1 deletions
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2
apps/app_dial.c
View File

@ -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;
}

6
channels/chan_sip.c
View File

@ -25347,6 +25347,12 @@ 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;