Small cleanup of handle_request_invite() - imported from 1.2 with changes

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47463 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 606657cfb208bccbff1373ccb8ca5d973e0fe90f..e36386dfd7101229e8a28f4532efdb8e2e0fd657 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -13100,14 +13100,13 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                }
 
                if (!replace_id && gotdest) {   /* No matching extension found */
-                       if (gotdest == 1 && ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
+                       if (gotdest == 1 && ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP))
                                transmit_response_reliable(p, "484 Address Incomplete", req);
-                               update_call_counter(p, DEC_CALL_LIMIT);
-                       } else {
+                       else
                                transmit_response_reliable(p, "404 Not Found", req);
-                               update_call_counter(p, DEC_CALL_LIMIT);
-                       }
+                       update_call_counter(p, DEC_CALL_LIMIT);
                        sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+                       return 0;
                } else {
                        /* If no extension was specified, use the s one */
                        /* Basically for calling to IP/Host name only */