[asterisk-commits] pabelanger: branch 1.4 r266579 - /branches/1.4/channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jun 1 09:54:14 CDT 2010
Author: pabelanger
Date: Tue Jun 1 09:54:05 2010
New Revision: 266579
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=266579
Log:
Missing fallback to audio fax feature when T.38 re-INVITE failed
When a T.38 re-INVITE failed with an 488 or 606 answer, we should
fallback to audio fax by send a re-re-INVITE without T.38. The
function is backported from 1.6 asterisk.
(closes issue #16795)
Reported by: vrban
(closes issue #16692)
Reported by: vrban
Patches:
t38_fallback_to_audio_v3.patch uploaded by vrban (license 756)
Tested by: lmadsen, vrban, haggard
https://reviewboard.asterisk.org/r/514/
Modified:
branches/1.4/channels/chan_sip.c
Modified: branches/1.4/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=266579&r1=266578&r2=266579
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Tue Jun 1 09:54:05 2010
@@ -13107,23 +13107,21 @@
case 606: /* Not Acceptable */
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 .
- */
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);
+ /* Trigger a reinvite back to audio */
+ transmit_reinvite_with_sdp(p);
+
+ if (p->owner && (p->owner->_state == AST_STATE_UP) && (bridgepeer = ast_bridged_channel(p->owner))) { /* if this is a re-invite */
+ struct sip_pvt *bridgepvt = NULL;
+ if (bridgepeer->tech == &sip_tech || bridgepeer->tech == &sip_tech_info) {
+ bridgepvt = (struct sip_pvt*)(bridgepeer->tech_pvt);
+ if (bridgepvt->udptl) {
+ sip_handle_t38_reinvite(bridgepeer, p, 0);
+ }
+ }
+ }
} else {
/* We can't set up this call, so give up */
if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
@@ -18980,7 +18978,7 @@
p->lastrtprx = p->lastrtptx = time(NULL);
ast_mutex_unlock(&p->lock);
return 0;
- } else { /* If we are handling sending 200 OK to the other side of the bridge */
+ } else if (pvt->t38.state != T38_DISABLED) { /* If we are handling sending 200 OK to the other side of the bridge */
if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE) && ast_test_flag(&pvt->flags[0], SIP_CAN_REINVITE)) {
ast_udptl_get_peer(pvt->udptl, &p->udptlredirip);
flag = 1;
@@ -19003,6 +19001,20 @@
p->lastrtprx = p->lastrtptx = time(NULL);
ast_mutex_unlock(&p->lock);
return 0;
+ } else if (pvt->t38.state == T38_DISABLED) { /* The other side can not talk T.38 with us. We tell it to the the originating T.38 party with a 488 */
+ p->t38.state = T38_DISABLED;
+ if (option_debug > 1) {
+ ast_log(LOG_DEBUG, "T38 changed state to %d on channel %s\n", pvt->t38.state, pvt->owner ? pvt->owner->name : "<none>");
+ ast_log(LOG_DEBUG, "T38 changed state to %d on channel %s\n", p->t38.state, chan ? chan->name : "<none>");
+ }
+ transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
+ p->lastrtprx = p->lastrtptx = time(NULL);
+ ast_mutex_unlock(&p->lock);
+ return 0;
+ } else {
+ ast_log(LOG_ERROR, "Something went wrong with T.38. State is:%d on channel %s and %d on channel %s\n", pvt->t38.state, pvt->owner ? pvt->owner->name : "<none>", p->t38.state, chan ? chan->name : "<none>");
+ ast_mutex_unlock(&p->lock);
+ return 0;
}
}
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