Merged revisions 65123 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
................
r65123 | oej | 2007-05-18 20:16:09 +0200 (Fri, 18 May 2007) | 10 lines
Merged revisions 65122 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 lines
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/trunk@65124
65c4cc65-6c06-0410-ace0-
fbb531ad65f3
const char *required;
unsigned int required_profile = 0;
struct ast_channel *c = NULL; /* New channel */
+ int reinvite = 0;
/* Find out what they support */
if (!p->sipoptions) {
else
ast_log(LOG_DEBUG, "Got a SIP re-transmit of INVITE for call %s\n", p->callid);
}
+ reinvite = 1;
c = p->owner;
}
}
/* Respond to normal re-invite */
if (sendok)
- transmit_response_with_sdp(p, "200 OK", req, ast_test_flag(req, SIP_PKT_IGNORE) ? XMIT_UNRELIABLE : XMIT_CRITICAL);
+ /* If this is not a re-invite or something to ignore - it's critical */
+ transmit_response_with_sdp(p, "200 OK", req, (reinvite || ast_test_flag(req, SIP_PKT_IGNORE)) ? XMIT_UNRELIABLE : XMIT_CRITICAL);
}
p->invitestate = INV_TERMINATED;
break;