]> git.ipfire.org Git - thirdparty/asterisk.git/commit
channels/chan_sip: Don't send a BYE after final response when PBX thread fails
authorMatthew Jordan <mjordan@digium.com>
Thu, 26 Feb 2015 03:02:24 +0000 (03:02 +0000)
committerMatthew Jordan <mjordan@digium.com>
Thu, 26 Feb 2015 03:02:24 +0000 (03:02 +0000)
commitbbfc8cc77816a6856ed01b45d7f7e9428d566372
tree063ff8eb3ab7cf01ef25157710997fb1403926f2
parenta0046c768cc397cd169edea3af05a454a157cab3
channels/chan_sip: Don't send a BYE after final response when PBX thread fails

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)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c