Merge from 1.4 : Don't send 183 reliably...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46383 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@ -3546,7 +3546,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
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if ((ast->_state != AST_STATE_UP) &&
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!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
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!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
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transmit_response_with_t38_sdp(p, "183 Session Progress", &p->initreq, XMIT_RELIABLE);
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transmit_response_with_t38_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
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ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
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}
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res = ast_udptl_write(p->udptl, frame);
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@ -12899,7 +12899,7 @@ static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, in
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/* We should answer something here. If we are here, the
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call we are replacing exists, so an accepted
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can't harm */
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transmit_response_with_sdp(p, "200 OK", req, 1);
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transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE);
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/* Do something more clever here */
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ast_channel_unlock(c);
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sip_pvt_unlock(p->refer->refer_call);
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@ -12908,7 +12908,7 @@ static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, in
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if (!c) {
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/* What to do if no channel ??? */
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ast_log(LOG_ERROR, "Unable to create new channel. Invite/replace failed.\n");
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transmit_response_with_sdp(p, "503 Service Unavailable", req, 1);
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transmit_response_reliable(p, "503 Service Unavailable", req);
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append_history(p, "Xfer", "INVITE/Replace Failed. No new channel.");
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sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
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sip_pvt_unlock(p->refer->refer_call);
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@ -12933,7 +12933,7 @@ static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, in
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Targetcall is not touched by the masq */
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/* Answer the incoming call and set channel to UP state */
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transmit_response_with_sdp(p, "200 OK", req, 1);
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transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE);
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ast_setstate(c, AST_STATE_UP);
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/* Stop music on hold and other generators */
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