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

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a6104e4da87e99fe7a2f6a3635b359c675256b3c..663f601e9f2f136ee8798784f4a6a966a2930bb1 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10593,6 +10593,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
        char *supported;
        char *required;
        unsigned int required_profile = 0;
+       int reinvite = 0;
 
        /* Find out what they support */
        if (!p->sipoptions) {
@@ -10733,6 +10734,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
        } else {
                if (option_debug > 1 && sipdebug)
                        ast_log(LOG_DEBUG, "Got a SIP re-invite for call %s\n", p->callid);
+               reinvite = 1;
                c = p->owner;
        }
        if (!ignore && p)
@@ -10809,7 +10811,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                        transmit_response(p, "180 Ringing", req);
                        break;
                case AST_STATE_UP:
-                       transmit_response_with_sdp(p, "200 OK", req, 1);
+                       /* If this is not a re-invite or something to ignore - it's critical */
+                       transmit_response_with_sdp(p, "200 OK", req, (ignore || reinvite) ? 1 : 2);
                        break;
                default:
                        ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", c->_state);