[asterisk-commits] oej: branch 1.4 r46382 -
/branches/1.4/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Sat Oct 28 12:08:59 MST 2006
Author: oej
Date: Sat Oct 28 14:08:58 2006
New Revision: 46382
URL: http://svn.digium.com/view/asterisk?rev=46382&view=rev
Log:
- 183 is not reliable message...
- Error should not have SDP
Modified:
branches/1.4/channels/chan_sip.c
Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?rev=46382&r1=46381&r2=46382&view=diff
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Sat Oct 28 14:08:58 2006
@@ -3458,7 +3458,7 @@
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
- transmit_response_with_t38_sdp(p, "183 Session Progress", &p->initreq, XMIT_RELIABLE);
+ transmit_response_with_t38_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
}
res = ast_udptl_write(p->udptl, frame);
@@ -12704,7 +12704,7 @@
/* We should answer something here. If we are here, the
call we are replacing exists, so an accepted
can't harm */
- transmit_response_with_sdp(p, "200 OK", req, 1);
+ transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE);
/* Do something more clever here */
ast_channel_unlock(c);
ast_mutex_unlock(&p->refer->refer_call->lock);
@@ -12713,7 +12713,7 @@
if (!c) {
/* What to do if no channel ??? */
ast_log(LOG_ERROR, "Unable to create new channel. Invite/replace failed.\n");
- transmit_response_with_sdp(p, "503 Service Unavailable", req, 1);
+ transmit_response_reliable(p, "503 Service Unavailable", req);
append_history(p, "Xfer", "INVITE/Replace Failed. No new channel.");
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_mutex_unlock(&p->refer->refer_call->lock);
@@ -12738,7 +12738,7 @@
Targetcall is not touched by the masq */
/* Answer the incoming call and set channel to UP state */
- transmit_response_with_sdp(p, "200 OK", req, 1);
+ transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE);
ast_setstate(c, AST_STATE_UP);
/* Stop music on hold and other generators */
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