[test-results] [Bamboo] Asterisk Testing > Asterisk Trunk > #342 has FAILED (11 tests failed, no failures were new). Change made by Mark Michelson.

[Bamboo]

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Asterisk Testing > Asterisk Trunk > #342 failed.
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Code has been updated by Mark Michelson.
11/249 tests failed, no failures were new.

http://bamboo.asterisk.org/browse/TESTING-ASTERISKTRUNK-342/


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Failing Jobs
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  - Asterisk CentOS 6 64-Bit (CentOS 6): 11 of 249 tests failed.



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Code Changes
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Mark Michelson (368637):

>Fix a specific scenario where ACKs are not matched.
>
>If a dialog-starting INVITE contains a to-tag, then Asterisk
>will respond with a 481. In this case, the resulting incoming
>ACK would not be matched, so Asterisk would continue retransmitting
>the 481 until the transaction times out.
>
>There were two issues. Asterisk, upon creating a sip_pvt would generate
>a local tag. However, when the time came to transmit the 481, since there
>was a to-tag in the INVITE, Asterisk would place this original to-tag
>in the 481 response. When the ACK came in, Asterisk would attempt to
>match the to-tag in the ACK to the generated local tag. Unfortunately,
>Asterisk never actually transmitted a response with the generated local
>tag, so the to-tag in the ACK would not match.
>
>The other problem was that when the 481 was sent, nothing was set
>on the sip_pvt to indicate what CSeq is expected in the ACK.
>
>To fix the first problem, we zero out the to-tag seen in the incoming
>INVITE. This way, Asterisk, when time to send a response, will send
>its generated local tag instead.
>
>To fix the second problem, we set the sip_pvt's pendinginvite to the
>CSeq of the INVITE when we send a 481.
>
>(closes issue ASTERISK-19892)
>Reported by Mark Michelson
>........
>
>Merged revisions 368625 from http://svn.asterisk.org/svn/asterisk/branches/1.8
>........
>
>Merged revisions 368629 from http://svn.asterisk.org/svn/asterisk/branches/10
>



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Tests
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Existing Test Failures (11)
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail forwarding
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail priority
   - AsteriskTestSuite: S/apps/voicemail/check voicemail options record unavail
   - AsteriskTestSuite: S/apps/voicemail/check voicemail new user
   - AsteriskTestSuite: S/queues/queue transfer callee
   - AsteriskTestSuite: S/apps/voicemail/check voicemail options record name
   - AsteriskTestSuite: S/apps/voicemail/check voicemail options record busy
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail forwarding auto urgent
   - AsteriskTestSuite: S/apps/voicemail/leave voicemail nominal
   - AsteriskTestSuite: S/apps/voicemail/check voicemail options record temp
   - AsteriskTestSuite: S/channels/ s i p/sip custom presence/non digium state change

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