Merged revisions 50006 via svnmerge from
[asterisk/asterisk.git] / channels / chan_sip.c
index 35a239d..56e9d70 100644 (file)
@@ -11966,6 +11966,33 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
                else if (!ast_test_flag(req, SIP_PKT_IGNORE))
                        update_call_counter(p, DEC_CALL_LIMIT);
                break;
+       case 488: /* Not acceptable here */
+               transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
+               if (reinvite && p->udptl) {
+                       /* If this is a T.38 call, we should go back to 
+                          audio. If this is an audio call - something went
+                          terribly wrong since we don't renegotiate codecs,
+                          only IP/port .
+                       */
+                       p->t38.state = T38_DISABLED;
+                       /* Try to reset RTP timers */
+                       ast_rtp_set_rtptimers_onhold(p->rtp);
+                       ast_log(LOG_ERROR, "Got error on T.38 re-invite. Bad configuration. Peer needs to have T.38 disabled.\n");
+
+                       /*! \bug Is there any way we can go back to the audio call on both
+                          sides here? 
+                       */
+                       /* While figuring that out, hangup the call */
+                       if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
+                               ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+                       ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);    
+               } else {
+                       /* We can't set up this call, so give up */
+                       if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
+                               ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
+                       ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);    
+               }
+               break;
        case 491: /* Pending */
                /* we really should have to wait a while, then retransmit */
                        /* We should support the retry-after at some point */
@@ -12404,6 +12431,10 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
                        if (sipmethod == SIP_INVITE)
                                handle_response_invite(p, resp, rest, req, seqno);
                        break;
+               case 488: /* Not acceptable here - codec error */
+                       if (sipmethod == SIP_INVITE)
+                               handle_response_invite(p, resp, rest, req, seqno);
+                       break;
                case 491: /* Pending */
                        if (sipmethod == SIP_INVITE)
                                handle_response_invite(p, resp, rest, req, seqno);
@@ -12460,7 +12491,6 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
                                                ast_string_field_build(p->owner, call_forward,
                                                                       "Local/%s@%s", p->username, p->context);
                                        /* Fall through */
-                               case 488: /* Not acceptable here - codec error */
                                case 480: /* Temporarily Unavailable */
                                case 404: /* Not Found */
                                case 410: /* Gone */