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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3736/">https://reviewboard.asterisk.org/r/3736/</a>
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<p style="margin-top: 0;">On July 10th, 2014, 11 a.m. CDT, <b>Corey Farrell</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;">Have you done leak testing? I suspect the change to __frame_free will result in leaks that will be reported by REF_DEBUG.</pre>
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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;">I have been watching for leaks here. There is nothing major that I see, but I have some further investigation to do.</pre>
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<p>- opticron</p>
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<p>On July 10th, 2014, 10:58 a.m. CDT, opticron wrote:</p>
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<div>Review request for Asterisk Developers, Corey Farrell, Joshua Colp, and Matt Jordan.</div>
<div>By opticron.</div>
<p style="color: grey;"><i>Updated July 10, 2014, 10:58 a.m.</i></p>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23960">ASTERISK-23960</a>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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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;">This contains the fixes necessary to get a PJSIP call up and playing files from the Playback() application with a translation path in the mix.
* channel.c: The set format helper functions were not actually setting the newly chosen formats resulting in no audio.
* frame.c: The format on the frame was being overzealously unreffed resulting in a crash.
* sorcery.c: The codec retrieval code was using the wrong level of indirection for ast_format_cap structures resuting in a crash for "pjsip show endpoint x"
* translate.c: The chosen codecs were being set backward on set vs native for what was actually desired causing incorrect codecs to be chosen.
* res_pjsip_sdp_rtp.c: An ast_rtp_codecs struct was not being initialized properly causing a crash.</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;">Call testing.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>team/group/media_formats-reviewed-trunk/res/res_pjsip_sdp_rtp.c <span style="color: grey">(418253)</span></li>
<li>team/group/media_formats-reviewed-trunk/main/translate.c <span style="color: grey">(418253)</span></li>
<li>team/group/media_formats-reviewed-trunk/main/sorcery.c <span style="color: grey">(418253)</span></li>
<li>team/group/media_formats-reviewed-trunk/main/frame.c <span style="color: grey">(418253)</span></li>
<li>team/group/media_formats-reviewed-trunk/main/channel.c <span style="color: grey">(418253)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3736/diff/" style="margin-left: 3em;">View Diff</a></p>
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