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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)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203116 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant 2009-06-25 16:04:10 +00:00
parent 80822297d4
commit c6a986222e
1 changed files with 20 additions and 16 deletions
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36
channels/chan_sip.c
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@ -20494,7 +20494,23 @@ 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);
@ -20555,21 +20571,9 @@ 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);