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/1.6.2@305253 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/apps/app_dial.c b/apps/app_dial.c
index 5feb39762cff37639b15edbdc43f213d1e0a81be..19782f64ae1afdf2dc791bb1fa4cd1d7bfa70e0e 100644 (file)
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -1671,7 +1671,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
                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;
                }

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index dce6f5ba5824064d184485d933a5eba732c5c0c2..03b021ae4196a488f08220f52867bd806370bdfa 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -23611,6 +23611,12 @@ static struct ast_channel *sip_request_call(const char *type, int format, void *
        }
        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;