remove some duplicated code;

fix indentation on one line;
mark XXX some unreachable code;
mark XXX another place where we could reduce the nesting depth.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@31872 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index e57942b8340a86c42f23c973734c68c9689569eb..0d1c828df3bfaa5f2c2df1528b80ae06cb3a6739 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -11222,6 +11222,8 @@ static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, str
                        ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);    
                return -1;
        } else {
+               /* XXX reduce nesting depth */
+
                /* Handle REFER notifications */
 
                char buf[1024];
@@ -11321,6 +11323,7 @@ static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, str
                return res;
        };
 
+       /* XXX hey, we never reach this code! */
        /* THis could be voicemail notification */
        transmit_response(p, "200 OK", req);
        if (!p->lastinvite) 
@@ -11370,7 +11373,7 @@ static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, in
                /* We have no bridge */
                if (!earlyreplace) {
                        if (option_debug > 1)
-                       ast_log(LOG_DEBUG, "    Attended transfer attempted to replace call with no bridge (maybe ringing). Channel %s!\n", replacecall->name);
+                               ast_log(LOG_DEBUG, "    Attended transfer attempted to replace call with no bridge (maybe ringing). Channel %s!\n", replacecall->name);
                        oneleggedreplace = 1;
                }
        } 
@@ -11935,20 +11938,19 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                }
        } else {
                if (p && !ast_test_flag(&p->flags[0], SIP_NEEDDESTROY)) {
-                       if (!p->jointcapability) {
-                               if (ast_test_flag(req, SIP_PKT_IGNORE))
-                                       transmit_response(p, "488 Not Acceptable Here (codec error)", req);
-                               else
-                                       transmit_response_reliable(p, "488 Not Acceptable Here (codec error)", req);
-                               ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);    
-                       } else {
+                       const char *msg;
+
+                       if (!p->jointcapability)
+                               msg = "488 Not Acceptable Here (codec error)";
+                       else {
                                ast_log(LOG_NOTICE, "Unable to create/find SIP channel for this INVITE\n");
-                               if (ast_test_flag(req, SIP_PKT_IGNORE))
-                                       transmit_response(p, "503 Unavailable", req);
-                               else
-                                       transmit_response_reliable(p, "503 Unavailable", req);
-                               ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);    
+                               msg = "503 Unavailable";
                        }
+                       if (ast_test_flag(req, SIP_PKT_IGNORE))
+                               transmit_response(p, msg, req);
+                       else
+                               transmit_response_reliable(p, msg, req);
+                       ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);    
                }
        }
        return res;