Merged revisions 203116 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

................
  r203116 | russell | 2009-06-25 11:04:10 -0500 (Thu, 25 Jun 2009) | 18 lines
  
  Merged revisions 203115 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r203115 | russell | 2009-06-25 11:02:16 -0500 (Thu, 25 Jun 2009) | 11 lines
    
    Resolve a crash related to a T.38 reinvite race condition.
    
    This change resolves a crash observed locally during some T.38 testing.
    A call was set up using a call file, and when the T.38 reinvite came in,
    the channel state was still AST_STATE_DOWN.  The reason is explained by
    a comment in the code that previously lived in the handling of
    AST_STATE_RINGING.  This change modifies the logic to handle the same
    race condition for any channel state that is not UP.
    
    (closes ABE-1895)
  ........
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@203119 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -19519,7 +19519,23 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 
 
 	if (c) {	/* We have a call  -either a new call or an old one (RE-INVITE) */
-		switch(c->_state) {
+		enum ast_channel_state c_state = c->_state;
+
+		if (c_state != AST_STATE_UP && reinvite &&
+			(p->invitestate == INV_TERMINATED || p->invitestate == INV_CONFIRMED)) {
+			/* If these conditions are true, and the channel is still in the 'ringing'
+			 * state, then this likely means that we have a situation where the initial
+			 * INVITE transaction has completed *but* the channel's state has not yet been
+			 * changed to UP. The reason this could happen is if the reinvite is received
+			 * on the SIP socket prior to an application calling ast_read on this channel
+			 * to read the answer frame we earlier queued on it. In this case, the reinvite
+			 * is completely legitimate so we need to handle this the same as if the channel 
+			 * were already UP. Thus we are purposely falling through to the AST_STATE_UP case.
+			 */
+			c_state = AST_STATE_UP;
+		}
+
+		switch(c_state) {
 		case AST_STATE_DOWN:
 			ast_debug(2, "%s: New call is still down.... Trying... \n", c->name);
 			transmit_response(p, "100 Trying", req);
@@ -19581,21 +19597,9 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 			p->invitestate = INV_PROCEEDING;
 			break;
 		case AST_STATE_RINGING:
-			if (reinvite && (p->invitestate == INV_TERMINATED || p->invitestate == INV_CONFIRMED)) {
-			/* If these conditions are true, and the channel is still in the 'ringing'
-			 * state, then this likely means that we have a situation where the initial
-			 * INVITE transaction has completed *but* the channel's state has not yet been
-			 * changed to UP. The reason this could happen is if the reinvite is received
-			 * on the SIP socket prior to an application calling ast_read on this channel
-			 * to read the answer frame we earlier queued on it. In this case, the reinvite
-			 * is completely legitimate so we need to handle this the same as if the channel 
-			 * were already UP. Thus we are purposely falling through to the AST_STATE_UP case.
-			 */
-			} else {
-				transmit_response(p, "180 Ringing", req);
-				p->invitestate = INV_PROCEEDING;
-				break;
-			}
+			transmit_response(p, "180 Ringing", req);
+			p->invitestate = INV_PROCEEDING;
+			break;
 		case AST_STATE_UP:
 			ast_debug(2, "%s: This call is UP.... \n", c->name);