diff options
| author | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-03-12 22:52:58 +0000 |
|---|---|---|
| committer | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-03-12 22:52:58 +0000 |
| commit | e9949822979f0b1c92d16c9af2e1ba683a5dc467 (patch) | |
| tree | 2d6a93db9c56ddf66ba6bad59c3ac03d6fb2b869 /channels | |
| parent | 4725920176ee0ba2f7eb5f567b4218358150c539 (diff) | |
Merged revisions 107157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r107157 | file | 2008-03-10 15:00:21 -0500 (Mon, 10 Mar 2008) | 4 lines
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: http://svn.digium.com/svn/asterisk/branches/1.6.0@108354 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 19 |
1 files changed, 5 insertions, 14 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 46fd9f10c..bb4136efa 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -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); @@ -14714,24 +14716,13 @@ 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. |