diff options
author | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-06-25 16:08:18 +0000 |
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committer | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-06-25 16:08:18 +0000 |
commit | ce0d40ec67e0cf8068d3cad5dac80a9eb3d3c136 (patch) | |
tree | f9282cb1317101208db14485272c8adfdf1b4452 | |
parent | 862e3049cd11f5b8cb1552394681d2a6bcf78896 (diff) |
Merged revisions 203116 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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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
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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: http://svn.digium.com/svn/asterisk/branches/1.6.2@203119 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r-- | channels/chan_sip.c | 36 |
1 files changed, 20 insertions, 16 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index e344940e2..363828eb1 100644 --- a/channels/chan_sip.c +++ b/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); |