[test-results] [Bamboo] Asterisk - 1.6.2 > Ubuntu Lucid (10.04) > #125 was SUCCESSFUL (with 92 tests). Change made by rmudgett.
Bamboo
bamboo at asterisk.org
Tue Jan 18 13:18:37 CST 2011
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Asterisk - 1.6.2 > Ubuntu Lucid (10.04) > #125 was successful.
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Code has been updated by rmudgett.
All 2 jobs passed with 92 tests in total.
http://bamboo.asterisk.org/browse/AST162-LUCID-125/
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Code Changes
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rmudgett (302173):
>Merged revisions 302172 via svnmerge from
>https://origsvn.digium.com/svn/asterisk/branches/1.4
>
>........
> r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) | 88 lines
>
> 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/
>........
>
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Tests
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Fixed Tests (1)
- AsteriskTestSuite: S/feature blonde transfer
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