Merged revisions 50006 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r50006 | oej | 2007-01-08 15:26:14 +0100 (Mon, 08 Jan 2007) | 11 lines

Issue #8677 - Handle failure of T.38 re-invite

This is not a fix, but adding an error message to tell the admin that
we have a bad configuration. We should not send T.38 re-invites to devices
that can't handle it (with the current architecture where you have to
hard-code t.38 support per device).

To really fix this, we need to figure out a way to tell the incoming
call that the re-invite failed, so we can signal failure on that
end and go back to the original call.

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50007 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 35a239d701a3c0ab7f6dd7aa86580729c2826006..56e9d70f86a08f9d0a46b86b05f264b2ca8dc1e4 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -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 */