From: Paul Belanger Date: Tue, 1 Jun 2010 14:54:05 +0000 (+0000) Subject: Missing fallback to audio fax feature when T.38 re-INVITE failed X-Git-Tag: 1.4.33-rc1~5 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=abc4bceec2471e79ff9794f2502963251e0a31fc;p=thirdparty%2Fasterisk.git 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@266579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 84209b0dd8..bcc3469a6e 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -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 : ""); + ast_log(LOG_DEBUG, "T38 changed state to %d on channel %s\n", p->t38.state, chan ? chan->name : ""); + } + 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 : "", p->t38.state, chan ? chan->name : ""); + ast_mutex_unlock(&p->lock); + return 0; } }