If we receive a 488 on a T38 request reinvite back to audio. As well reinvite across...

(closes issue #8677)
Reported by: alex-911

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107157 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a4afae1a90110c3e389e75bfc2979f71d2eada17..39b2d27ea36adf0b49c41e86d7acdf27ad2cd3f6 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5133,6 +5133,8 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
                                        transmit_reinvite_with_sdp(p, TRUE, FALSE);
                                }
                                break;
+                       case AST_T38_TERMINATED:
+                       case AST_T38_REFUSED:
                        case AST_T38_REQUEST_TERMINATE:         /* Shutdown T38 */
                                if (p->t38.state == T38_ENABLED)
                                        transmit_reinvite_with_sdp(p, FALSE, FALSE);
@@ -14692,24 +14694,13 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
                break;
        case 488: /* Not acceptable here */
                xmitres = 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 .
-                       */
+               if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
                        change_t38_state(p, 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 && !req->ignore)
-                               ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
-                       p->needdestroy = 1;
+                       /* Trigger a reinvite back to audio */
+                       transmit_reinvite_with_sdp(p, FALSE, FALSE);
                } else if (p->udptl && p->t38.state == T38_LOCAL_DIRECT) {
                        /* We tried to send T.38 out in an initial INVITE and the remote side rejected it,
                           right now we can't fall back to audio so totally abort.