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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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<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; "> <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; "> <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">""</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; "> <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="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 locks around this stream_and_wait need to go. They stopped the choppy audio, which happens with DAHDI and SIP, seems as though it'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>- Alec</p>
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<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't be playing out the fail/success audio, as it has 'netlock' locked.
2). dialplan applications for directed_pickups shouldn'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 => 71,1,Pickup() ; any ringing extension in same pickupgroup
exten => 72,1,Pickup(85@phones) ; dahdi extension
exten => 73,1,Pickup(823@phones) ; sip extension
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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=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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