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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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<a href="https://reviewboard.asterisk.org/r/1047/diff/1/?file=14324#file14324line1586" style="color: black; font-weight: bold; text-decoration: underline;">/branches/1.6.2/main/features.c</a>
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(Diff revision 1)
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int builtin_atxfer(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, char *code, int sense, void *data)</pre></td>
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<th bgcolor="#e9eaa8" style="border-right: 1px solid #C0C0C0;" align="right"><font size="2">1484</font></th>
<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">        <span class="n">newchan</span> <span class="o">=</span> <span class="n">feature_request_and_dial</span><span class="p">(</span><span class="n">transferer</span><span class="p">,</span> <span class="n">transferee</span><span class="p">,</span> <span class="s">"Local"</span><span class="p">,</span> <span class="n">ast_best_codec</span><span class="p">(</span><span class="n">transferer</span><span class="o">-></span><span class="n">nativeformats</span><span class="p">),</span></pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="cm">/* BUGBUG need to make DAHDI/i4/400 a dialable channel name on 1.8. */</span></pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">The three BUGBUG comments in this diff deal with porting this patch to 1.8. They will be removed when the patch is committed.</pre>
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<p>- rmudgett</p>
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<p>On December 3rd, 2010, 6:44 p.m., rmudgett wrote:</p>
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<div>Review request for Asterisk Developers and Russell Bryant.</div>
<div>By rmudgett.</div>
<p style="color: grey;"><i>Updated 2010-12-03 18:44:42</i></p>
<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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