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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3245/">https://reviewboard.asterisk.org/r/3245/</a>
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<a href="https://reviewboard.asterisk.org/r/3245/diff/1/?file=54263#file54263line87" style="color: black; font-weight: bold; text-decoration: underline;">branches/12/res/res_pjsip_send_to_voicemail.c</a>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="kt">int</span> <span class="n">size</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;">You don't need to pass size in, it's optional.</pre>
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<a href="https://reviewboard.asterisk.org/r/3245/diff/1/?file=54263#file54263line124" style="color: black; font-weight: bold; text-decoration: underline;">branches/12/res/res_pjsip_send_to_voicemail.c</a>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="n">RAII_VAR</span><span class="p">(</span><span class="k">struct</span> <span class="n">ast_datastore</span> <span class="o">*</span><span class="p">,</span> <span class="n">sip_session_datastore</span><span class="p">,</span> <span class="nb">NULL</span><span class="p">,</span> <span class="n">ao2_cleanup</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;">Silly use of RAII_VAR.</pre>
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<a href="https://reviewboard.asterisk.org/r/3245/diff/1/?file=54263#file54263line141" style="color: black; font-weight: bold; text-decoration: underline;">branches/12/res/res_pjsip_send_to_voicemail.c</a>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="n">ast_log</span><span class="p">(</span><span class="n">LOG_WARNING</span><span class="p">,</span> <span class="s">"%s attempted to transfer to voicemail, "</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span><span class="s">"but was not in a two party bridge.</span><span class="se">\n</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="tb"> </span><span class="tb"> </span><span class="tb"> </span><span class="n">ast_channel_name</span><span class="p">(</span><span class="n">session</span><span class="o">-></span><span class="n">channel</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;">I think it would also be useful to have the endpoint name in here as well.</pre>
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<a href="https://reviewboard.asterisk.org/r/3245/diff/1/?file=54263#file54263line151" style="color: black; font-weight: bold; text-decoration: underline;">branches/12/res/res_pjsip_send_to_voicemail.c</a>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="n">ast_channel_cleanup</span><span class="p">(</span><span class="n">other_party</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;">No need for ast_channel_cleanup, just use ast_channel_unref.</pre>
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<a href="https://reviewboard.asterisk.org/r/3245/diff/1/?file=54263#file54263line179" style="color: black; font-weight: bold; text-decoration: underline;">branches/12/res/res_pjsip_send_to_voicemail.c</a>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="n">RAII_VAR</span><span class="p">(</span><span class="k">struct</span> <span class="n">ast_datastore</span> <span class="o">*</span><span class="p">,</span> <span class="n">feature_datastore</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="tb"> </span><span class="tb"> </span><span class="n">ast_sip_session_get_datastore</span><span class="p">(</span><span class="n">session</span><span class="p">,</span> <span class="n">DATASTORE_NAME</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="tb"> </span><span class="tb"> </span><span class="n">ao2_cleanup</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;">Pfft, using RAII_VAR here is silly.</pre>
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<p>- Joshua Colp</p>
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<p>On February 20th, 2014, 11:01 p.m. UTC, Kevin Harwell wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Kevin Harwell.</div>
<p style="color: grey;"><i>Updated Feb. 20, 2014, 11:01 p.m.</i></p>
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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;">Added the ability for transferring directly to voicemail on digium phones. Added a new module that checks for the presence of a custom header and/or diversion header within a sip REFER. If either is found and they specify a sending to voicemail action then variables are added to the channel allowing the user access to them in the dialplan. Dialplan can then be written that branches based upon these values allowing, for instace, for a single number to be used for dialing and/or accessing voicemail directly.
Also fixed a problem where the PJSIP_HEADER function was allowing non pjsip channels through (checked to make sure it has the correct channel type before proceeding).</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;">Ran various scenarios manually with digium phones to make sure user were able to transfer callers directly to voicemail. Also wrote a testsuite test that checks the presence of those headers/values in the dialplan: https://reviewboard.asterisk.org/r/3246/</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>branches/12/res/res_pjsip_send_to_voicemail.c <span style="color: grey">(PRE-CREATION)</span></li>
<li>branches/12/res/res_pjsip_header_funcs.c <span style="color: grey">(408442)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3245/diff/" style="margin-left: 3em;">View Diff</a></p>
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