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If we receive a 488 on a T38 request reinvite back to audio. As well reinvite across a bridge back to audio if one side doesn't negotiate to T38. 

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp 2008-03-10 20:00:21 +00:00
parent 86fe5cc800
commit 362b184c9c
1 changed files with 5 additions and 14 deletions
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19
channels/chan_sip.c
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@ -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.