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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1047/">https://reviewboard.asterisk.org/r/1047/</a>
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<div>Review request for Asterisk Developers and Russell Bryant.</div>
<div>By rmudgett.</div>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Issue 17999
1) A (8123364000) calls B (0011*102). B answers.
2) B using DTMF dial *2 (code in features.conf for attendant transfer).
3) A hears MOH. B dial number C (3364021)
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 rining (no aswer)
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.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">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 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)
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.</pre>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/view.php?id=17273">17273</a>,
<a href="https://issues.asterisk.org/view.php?id=17999">17999</a>,
<a href="https://issues.asterisk.org/view.php?id=18395">18395</a>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/1.6.2/main/features.c <span style="color: grey">(297577)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1047/diff/" style="margin-left: 3em;">View Diff</a></p>
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