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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3990/
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(Updated Sept. 18, 2014, 5:50 p.m.)
Review request for Asterisk Developers, Matt Jordan and rmudgett.
Changes
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Fix reference leak rmudgett pointed out.
Bugs: ASTERISK-24237
https://issues.asterisk.org/jira/browse/ASTERISK-24237
Repository: Asterisk
Description
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Reproduction:
pj123 calls 1601
pj123 does a SIP blonde transfer to 1603
1603 answers
FRACK occurs
all are PJSIP endpoints.
Basically what happens is there is a second outbound dial from another pj123
channel. Before the dial is answered, the pj123 gets masqueraded out of the
picture and replaced with a channel that isn't in the dial state.
This patch fixes that by advancing the incoming channel to the dial state in
the channel breakdown function of a datastore on the pj123 channel. Honestly,
I'm not sure this is entirely adequate, but it seems to work in all of the
conditions I've tried so far and it doesn't impede normal attended transfers. I
might need to try and figure out what happens if I masquerade in a channel that
is already dialing though. I'm not sure if that's something we can really
expect to happen under normal conditions, but that seems like something that
would screw up this approach.
Note that I think this patch is the right idea, I just don't know if I need to
account for the possibility that the channel that masquerades into pj123's
dialing channel might not be in the neutral state.
Diffs (updated)
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/branches/12/main/stasis_channels.c 422882
Diff: https://reviewboard.asterisk.org/r/3990/diff/
Testing
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Ran against reproduction of the above scenario. Assertion was gone and the CDR
results were as follows:
"","123","1601","default","""""
<123>","PJSIP/pj123-00000000","PJSIP/1601-00000001","Dial","PJSIP/1601,,tT","2014-09-11
21:48:51","2014-09-11 21:48:53","2014-09-11
21:48:57",5,4,"ANSWERED","DOCUMENTATION","1410472131.0",""
"","123","","default","""""
<123>","PJSIP/pj123-00000002","PJSIP/1603-00000003","Dial","PJSIP/1603","2014-09-11
21:48:55",,"2014-09-11 21:48:57",2,0,"NO
ANSWER","DOCUMENTATION","1410472135.6",""
"","1601","1603","default","""""
<1601>","PJSIP/1601-00000001","PJSIP/1603-00000003","AppDial","(Outgoing
Line)","2014-09-11 21:48:57","2014-09-11 21:48:57","2014-09-11
21:49:04",6,6,"ANSWERED","DOCUMENTATION","1410472131.1",""
And dial events reported by AMI:
http://pastebin.com/tWuwL7xa
(note that there is a mismatch between the number of dial end and dial begin
events... not sure if this is a problem, but I might be able to fix it just by
updating the old chan, not sure what status code to use though).
Ran against normal attended transfer, feature attended transfers, and blind
transfers with no noticeable effect.
Ran against testsuite blonde transfer tests, some attended transfer tests, some
blind transfer tests, and all pjsip transfer tests (at least ones that will run
on my box... a few won't due to sipp version requirements that I really need to
get around to fixing eventually) for comparison purposes. All passed exhibiting
the same behavior as before as far as warning messages and such go.
Thanks,
Jonathan Rose
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