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author | Matthew Jordan <mjordan@digium.com> | 2015-02-26 03:03:39 +0000 |
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committer | Matthew Jordan <mjordan@digium.com> | 2015-02-26 03:03:39 +0000 |
commit | 3725173b9e18374e84af2fed59c245d5d15eb4bb (patch) | |
tree | a947c290eee1f709dae80c06b50683fcb5b0f0f5 /build_tools | |
parent | e484140aedda47d5f63f28a12ce776c34eedd066 (diff) |
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)
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Merged revisions 432320 from http://svn.asterisk.org/svn/asterisk/branches/11
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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
Diffstat (limited to 'build_tools')
0 files changed, 0 insertions, 0 deletions