channels/chan_sip: Don't send a BYE after final response when PBX thread fails
authorMatthew Jordan <mjordan@digium.com>
Thu, 26 Feb 2015 03:03:39 +0000 (03:03 +0000)
committerMatthew Jordan <mjordan@digium.com>
Thu, 26 Feb 2015 03:03:39 +0000 (03:03 +0000)
commit3725173b9e18374e84af2fed59c245d5d15eb4bb
treea947c290eee1f709dae80c06b50683fcb5b0f0f5
parente484140aedda47d5f63f28a12ce776c34eedd066
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)
........

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

Merged revisions 432321 from http://svn.asterisk.org/svn/asterisk/branches/13

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432322 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c