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channels/chan_sip: Don't send a BYE after final response when PBX thread fails

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When Asterisk fails to start a PBX thread for a new channel - for example, when
the maxcalls setting in asterisk.conf is exceeded - we currently send a final
response, and then attempt to send a BYE request to the UA. Since that's all
sorts of wrong, this patch fixes that by setting sipalreadygone on the sip_pvt
such that we don't get stuck sending BYE requests to something that does not
want it.

Note that this patch is a slight modification of the one on ASTERISK-15434.
For clarity, it explicitly calls sipalreadygone with the calls to transmit a
final response.

ASTERISK-21845
ASTERISK-15434 #close
Reported by: Makoto Dei
Tested by: Matt Jordan
patches:
  sip-pbxstart-failed.patch uploaded by Makoto Dei (License 5027)
........

Merged revisions 432320 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Matthew Jordan
2015-02-26 03:03:06 +00:00
parent 53aec7a969
commit 34989bd9c8
1 changed files with 2 additions and 0 deletions
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2
channels/chan_sip.c
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@@ -25806,11 +25806,13 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str
switch(result) {
case AST_PBX_FAILED:
sip_alreadygone(p);
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
p->invitestate = INV_COMPLETED;
transmit_response_reliable(p, "503 Unavailable", req);
break;
case AST_PBX_CALL_LIMIT:
sip_alreadygone(p);
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
p->invitestate = INV_COMPLETED;
transmit_response_reliable(p, "480 Temporarily Unavailable", req);