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

[rmudgett]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1047/
-----------------------------------------------------------

(Updated 2011-01-14 17:13:38.468989)


Review request for Asterisk Developers and Russell Bryant.


Changes
-------

Updated patch.
1) Made NULL tech code conditional with a big comment describing it and its consequences.
2) Fixed feature_request_and_dial() to defer frames for the transferee like autoservice.
3) Committed the unrelated wrong variable fix.


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 (updated)
-----

  /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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20110114/68a411b2/attachment-0001.htm>