make chan_sip able to deal with PBX-level call limit being reached (issue #5131)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6788 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index fee4848bb01f6894d16cc3e3bf8d7a060a7b80d8..3f338cfc54697751ed01d6f9a265b50eda76f217 100755 (executable)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10344,17 +10344,37 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                        transmit_response(p, "100 Trying", req);
                        ast_setstate(c, AST_STATE_RING);
                        if (strcmp(p->exten, ast_pickup_ext())) {
-                               if (ast_pbx_start(c)) {
+                               enum ast_pbx_result res;
+
+                               res = ast_pbx_start(c);
+
+                               switch (res) {
+                               case AST_PBX_FAILED:
+                                       ast_log(LOG_WARNING, "Failed to start PBX :(\n");
+                                       if (ignore)
+                                               transmit_response(p, "503 Unavailable", req);
+                                       else
+                                               transmit_response_reliable(p, "503 Unavailable", req, 1);
+                                       break;
+                               case AST_PBX_CALL_LIMIT:
+                                       ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
+                                       if (ignore)
+                                               transmit_response(p, "480 Temporarily Unavailable", req);
+                                       else
+                                               transmit_response_reliable(p, "480 Temporarily Unavailable", req, 1);
+                                       break;
+                               case AST_PBX_SUCCESS:
+                                       /* nothing to do */
+                                       break;
+                               }
+
+                               if (res) {
                                        ast_log(LOG_WARNING, "Failed to start PBX :(\n");
                                        /* Unlock locks so ast_hangup can do its magic */
                                        ast_mutex_unlock(&c->lock);
                                        ast_mutex_unlock(&p->lock);
                                        ast_hangup(c);
                                        ast_mutex_lock(&p->lock);
-                                       if (ignore)
-                                               transmit_response(p, "503 Unavailable", req);
-                                       else
-                                               transmit_response_reliable(p, "503 Unavailable", req, 1);
                                        c = NULL;
                                }
                        } else {