[test-results] [Bamboo] No agents to build plan Asterisk - 1.4 - FreeBSD 8.1 - i386

[Bamboo]

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AST14-FREEBSD81-I386-72 has been queued, but there's no agent capable of building it.
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http://bamboo.asterisk.org/browse/AST14-FREEBSD81-I386/log

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Code Changes
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rmudgett (302172):

>Issues with DTMF triggered attended transfers.
>
>Issue #17999
>1) A calls B. B answers.
>2) B using DTMF dial *2 (code in features.conf for attended transfer).
>3) A hears MOH. B dial number C
>4) C ringing. A hears MOH.
>5) B hangup. A still hears MOH. C ringing.
>6) A hangup. C still ringing until "atxfernoanswertimeout" expires.
>For v1.4 C will ring forever until C answers the dead line. (Issue #17096)
>
>Problem: When A and B hangup, C is still ringing.
>
>Issue #18395
>SIP call limit of B is 1
>1. A call B, B answered
>2. B *2(atxfer) call C
>3. B hangup, C ringing
>4. Timeout waiting for C to answer
>5. Recall to B fails because B has reached its call limit.
>
>Because B reached its call limit, it cannot do anything until the transfer
>it started completes.
>
>Issue #17273
>Same scenario as issue 18395 but party B is an FXS port.  Party B cannot
>do anything until the transfer it started completes.  If B goes back off
>hook before C answers, B hears ringback instead of the expected dialtone.
>
>**********
>Note for the issue #17273 and #18395 fix:
>
>DTMF attended transfer works within the channel bridge.  Unfortunately,
>when either party A or B in the channel bridge hangs up, that channel is
>not completely hung up until the transfer completes.  This is a real
>problem depending upon the channel technology involved.
>
>For chan_dahdi, the channel is crippled until the hangup is complete.
>Either the channel is not useable (analog) or the protocol disconnect
>messages are held up (PRI/BRI/SS7) and the media is not released.
>
>For chan_sip, a call limit of one is going to block that endpoint from any
>further calls until the hangup is complete.
>
>For party A this is a minor problem.  The party A channel will only be in
>this condition while party B is dialing and when party B and C are
>conferring.  The conversation between party B and C is expected to be a
>short one.  Party B is either asking a question of party C or announcing
>party A.  Also party A does not have much incentive to hangup at this
>point.
>
>For party B this can be a major problem during a blonde transfer.  (A
>blonde transfer is our term for an attended transfer that is converted
>into a blind transfer.  :)) Party B could be the operator.  When party B
>hangs up, he assumes that he is out of the original call entirely.  The
>party B channel will be in this condition while party C is ringing, while
>attempting to recall party B, and while waiting between call attempts.
>
>WARNING:
>The ATXFER_NULL_TECH conditional is a hack to fix the problem.  It will
>replace the party B channel technology with a NULL channel driver to
>complete hanging up the party B channel technology.  The consequences of
>this code is that the 'h' extension will not be able to access any channel
>technology specific information like SIP statistics for the call.
>
>ATXFER_NULL_TECH is not defined by default.
>**********
>
>(closes issue #17999)
>Reported by: iskatel
>Tested by: rmudgett
>JIRA SWP-2246
>
>(closes issue #17096)
>Reported by: gelo
>Tested by: rmudgett
>JIRA SWP-1192
>
>(closes issue #18395)
>Reported by: shihchuan
>Tested by: rmudgett
>
>(closes issue #17273)
>Reported by: grecco
>Tested by: rmudgett
>
>Review: https://reviewboard.asterisk.org/r/1047/
>

mnicholson (302311):

>URI encode the user part of the contact header.
>
>ABE-2705
>

tilghman (302663):

>Add some API documentation
>


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