From 5d4f272a90fb7a1bf10238f75dae2cab0c4f6eb1 Mon Sep 17 00:00:00 2001
From: Olle Johansson <oej@edvina.net>
Date: Mon, 8 Jan 2007 14:31:16 +0000
Subject: [PATCH] Merged revisions 50006 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r50006 | oej | 2007-01-08 15:26:14 +0100 (Mon, 08 Jan 2007) | 11 lines
Issue #8677 - Handle failure of T.38 re-invite
This is not a fix, but adding an error message to tell the admin that
we have a bad configuration. We should not send T.38 re-invites to devices
that can't handle it (with the current architecture where you have to
hard-code t.38 support per device).
To really fix this, we need to figure out a way to tell the incoming
call that the re-invite failed, so we can signal failure on that
end and go back to the original call.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_sip.c | 32 +++++++++++++++++++++++++++++++-
1 file changed, 31 insertions(+), 1 deletion(-)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 35a239d701..56e9d70f86 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -11966,6 +11966,33 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
else if (!ast_test_flag(req, SIP_PKT_IGNORE))
update_call_counter(p, DEC_CALL_LIMIT);
break;
+ case 488: /* Not acceptable here */
+ 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);
+ } else {
+ /* We can't set up this call, so give up */
+ 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);
+ }
+ break;
case 491: /* Pending */
/* we really should have to wait a while, then retransmit */
/* We should support the retry-after at some point */
@@ -12404,6 +12431,10 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
break;
+ case 488: /* Not acceptable here - codec error */
+ if (sipmethod == SIP_INVITE)
+ handle_response_invite(p, resp, rest, req, seqno);
+ break;
case 491: /* Pending */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
@@ -12460,7 +12491,6 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
ast_string_field_build(p->owner, call_forward,
"Local/%s@%s", p->username, p->context);
/* Fall through */
- case 488: /* Not acceptable here - codec error */
case 480: /* Temporarily Unavailable */
case 404: /* Not Found */
case 410: /* Gone */
--
GitLab