[asterisk-dev] [Code Review] Issues with DTMF triggered attended transfers.

[Russell Bryant]

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Ship it!



/branches/1.6.2/main/features.c
<https://reviewboard.asterisk.org/r/1047/#comment6359>

    s/useable/usable/



/branches/1.6.2/main/features.c
<https://reviewboard.asterisk.org/r/1047/#comment6360>

    I think we should probably change the reference to "blonde transfer" here.  I'm not sure how many people are going to know what that means.  ;-)


- Russell


On 2011-01-14 20:09:44, rmudgett wrote:
> 
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> 
> (Updated 2011-01-14 20:09:44)
> 
> 
> Review request for Asterisk Developers and Russell Bryant.
> 
> 
> Summary
> -------
> 
> 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.
> 5) A hangup. C still ringing until "atxfernoanswertimeout" expires.
> 
> 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, C ringing (no answer)
> 3. B hangup
> 4. C cancel call
> 5. Call 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.
> 
> 
> This addresses bugs 17273, 17999 and 18395.
>     https://issues.asterisk.org/view.php?id=17273
>     https://issues.asterisk.org/view.php?id=17999
>     https://issues.asterisk.org/view.php?id=18395
> 
> 
> Diffs
> -----
> 
>   /branches/1.6.2/main/features.c 302003 
> 
> Diff: https://reviewboard.asterisk.org/r/1047/diff
> 
> 
> Testing
> -------
> 
> Party A - transferee
> Party B - transferer
> Party C - target of transfer
> 
> A and B are connected (It does not matter who called whom for these tests.)
> B requests attended transfer feature by dialing *2 feature.
> 
> B fails to dial party C (Check A & B audio)
> B dials wrong number (Check A & B audio)
> B cancels call to party C with '*' (Check A & B audio)
> C is the parking extension (Outside the scope of this patch)
> C does not answer before timeout (Check A & B audio)
> C is busy (Check A & B audio)
> A hangs up before C answers (Check if A is completely released)
> 	(If A is an analog port it is dead until the user
> 	 configured xferfailsound completes playing.)
> 	(Test case is issue 17273 and issue 18395 related)
> C answers before B hangup (Attended transfer)
> 	A still online
> 		C hangs up first (Check A & B audio)
> 		B hangs up first (Check A & C audio)
> 	A hangs up (Check if A is completely released)
> 		(If A is an analog port it is dead until B or C hangs up)
> 		(Test case is issue 17273 and issue 18395 related)
> 		C hangs up first (Check B audio)
> 		B hangs up first (Check C audio)
> B hangs up before C answers (Blonde transfer) (Check if B is completely released)
> 	(Test case is issue 17273 and issue 18395)
> 	A hangs up (C should quit ringing immediately)
> 		(Test case is issue 17999)
> 	C answers (Check A & C audio)
> 	C does not answer before timeout
> 		A hangs up when B redialed (B should quit ringing immediately)
> 			(Test case is issue 17999 related)
> 		B answers recall (Check A & B audio)
> 		A hangs up when sleeping before redialing C (A should be released immediately)
> 			(Test case is issue 17999 related)
> 		A hangs up when C redialed (C should quit ringing immediately)
> 			(Test case is issue 17999 related)
> 		C answers recall (Check A & C audio)
> 		Noone answers (Check A audio)
> 
> Tests passed with exceptions noted.
> 
> 
> Thanks,
> 
> rmudgett
> 
>

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