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


<p style="color: grey;"><i>Updated 2011-05-11 19:25:03.501049</i></p>



<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Changes</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;">&#39;final&#39; tweaks as suggested.
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<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
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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> (updated)</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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