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Not getting an ACK to a 200 OK in the initial invite is critical to the call.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2@65122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -10593,6 +10593,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
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char *supported;
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char *required;
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unsigned int required_profile = 0;
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int reinvite = 0;
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/* Find out what they support */
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if (!p->sipoptions) {
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@@ -10733,6 +10734,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
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} else {
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if (option_debug > 1 && sipdebug)
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ast_log(LOG_DEBUG, "Got a SIP re-invite for call %s\n", p->callid);
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reinvite = 1;
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c = p->owner;
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}
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if (!ignore && p)
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@@ -10809,7 +10811,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
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transmit_response(p, "180 Ringing", req);
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break;
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case AST_STATE_UP:
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transmit_response_with_sdp(p, "200 OK", req, 1);
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/* If this is not a re-invite or something to ignore - it's critical */
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transmit_response_with_sdp(p, "200 OK", req, (ignore || reinvite) ? 1 : 2);
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break;
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default:
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ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", c->_state);
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