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Missing fallback to audio fax feature when T.38 re-INVITE failed

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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/



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@266579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Paul Belanger
2010-06-01 14:54:05 +00:00
parent 06914c13cf
commit abc4bceec2
1 changed files with 26 additions and 14 deletions
Download Patch File Download Diff File Expand all files Collapse all files
channels/chan_sip.c
+26 -14
View File
@@ -13107,23 +13107,21 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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");
/* Trigger a reinvite back to audio */
transmit_reinvite_with_sdp(p);
/*! \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);
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 @@ static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt
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 @@ static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt
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;
}
}