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      This is an automatically generated e-mail. To reply, visit:
      <a href="https://reviewboard.asterisk.org/r/1185/">https://reviewboard.asterisk.org/r/1185/</a>
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<blockquote style="margin-left: 1em; border-left: 2px solid #d0d0d0; padding-left: 10px;">
 <p style="margin-top: 0;">On May 13th, 2011, 6:10 p.m., <b>Alec Davis</b> wrote:</p>
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    <a href="https://reviewboard.asterisk.org/r/1185/diff/16/?file=16493#file16493line5740" style="color: black; font-weight: bold; text-decoration: underline;">trunk/main/features.c</a>
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     (Diff revision 16)

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   <td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int find_channel_by_group(void *obj, void *arg, void *data, int flags)</pre></td>

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    <td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "></pre></td>
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    <td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">                                                <span class="n">ast_channel_lock</span><span class="p">(</span><span class="n">target</span><span class="p">);</span></pre></td>
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    <td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "></pre></td>
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    <td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">                                                <span class="n">ast_stream_and_wait</span><span class="p">(</span><span class="n">target</span><span class="p">,</span> <span class="n">pickupsound</span><span class="p">,</span> <span class="s">&quot;&quot;</span><span class="p">);</span></pre></td>
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    <td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "></pre></td>
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    <td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">                                                <span class="n">ast_channel_unlock</span><span class="p">(</span><span class="n">target</span><span class="p">);</span></pre></td>
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  <pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">the locks around this stream_and_wait need to go. They stopped the choppy audio, which happens with DAHDI and SIP, seems as though it&#39;s an unrelated issue, induced by Pickup.

Using tt_monkeys to exagerate to problem, no other channels are able to dial out.</pre>
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 <p>On May 13th, 2011, 7:31 p.m., <b>rmudgett</b> wrote:</p>
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  <pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">The locks are needed because we are injecting the pickupsound outside the thread running the bridge.  Without the locks on my setup, the calls got disconnected because chan_dahdi complained about incompatible audio frames.  We need to get the bridge to inject the pickupsound for us somehow.

A follow-up patch is needed.</pre>
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</blockquote>
<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">This review needs to be closed.  It has been submitted.  Another issue needs to be created for the held lock problem identified above.</pre>
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<p>- rmudgett</p>


<br />
<p>On May 13th, 2011, 6:09 p.m., Alec Davis wrote:</p>






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<div>Review request for Asterisk Developers and rmudgett.</div>
<div>By Alec Davis.</div>


<p style="color: grey;"><i>Updated 2011-05-13 18:09:58</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;">Since 1.8, the new pickupsound and pickupfailsound in features.conf cause many issues.

1). chan_sip:handle_request_invite() shouldn&#39;t be playing out the fail/success audio, as it has &#39;netlock&#39; locked.
2). dialplan applications for directed_pickups shouldn&#39;t beep.
3). feature code for directed pickup should beep on success/failure if configured.

Moved app_directed:pickup_do() to features:ast_do_pickup().

Functions below, all now use the new ast_do_pickup()
app_directed_pickup.c:
   pickup_by_channel()
   pickup_by_exten()
   pickup_by_mark()
   pickup_by_part()
features.c:
   ast_pickup_call()

Created a sip_pickup() thread to handle the pickup and playout the audio, spawned from handle_request_invite.</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;">pickup using *8 feature code, with pickup sounds enabled/disabled

exten =&gt; 71,1,Pickup()           ; any ringing extension in same pickupgroup 
exten =&gt; 72,1,Pickup(85@phones)  ; dahdi extension
exten =&gt; 73,1,Pickup(823@phones) ; sip extension
</pre>
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<div style="margin-top: 1.5em;">
 <b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>


 <a href="https://issues.asterisk.org/view.php?id=18654">18654</a>


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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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 <li>trunk/apps/app_directed_pickup.c <span style="color: grey">(317665)</span></li>

 <li>trunk/channels/chan_sip.c <span style="color: grey">(317665)</span></li>

 <li>trunk/include/asterisk/features.h <span style="color: grey">(317665)</span></li>

 <li>trunk/main/features.c <span style="color: grey">(317665)</span></li>

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<p><a href="https://reviewboard.asterisk.org/r/1185/diff/" style="margin-left: 3em;">View Diff</a></p>




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