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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1199/">https://reviewboard.asterisk.org/r/1199/</a>
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<a href="https://reviewboard.asterisk.org/r/1199/diff/1/?file=16256#file16256line5381" style="color: black; font-weight: bold; text-decoration: underline;">/branches/1.8/channels/chan_sip.c</a>
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(Diff revision 1)
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int sip_call(struct ast_channel *ast, char *dest, int timeout)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="k">if</span> <span class="p">(</span><span class="o">!</span><span class="n">ast_strlen_zero</span><span class="p">(</span><span class="n">p</span><span class="o">-></span><span class="n">cid_num</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="o">||</span> <span class="p">(</span><span class="n">p</span><span class="o">-></span><span class="n">callingpres</span> <span class="o">&</span> <span class="n">AST_PRES_RESTRICTION</span><span class="p">)</span> <span class="o">!=</span> <span class="n">AST_PRES_ALLOWED</span><span class="p">)</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;">Can you explain this if statement? It reads a bit weird. This reads to me like you're going to update the connected number if either there is a cid_num set for p or if the calling presentation is not allowed. It seems like it should only be updating the connected number if there is a cid_num set for p AND the calling presentation IS allowed.
I'm probably just misinterpreting it, but a comment in the code would help to make it less confusing.</pre>
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<p>- Mark</p>
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<p>On April 26th, 2011, 5:49 p.m., rmudgett wrote:</p>
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<div>Review request for Asterisk Developers and Mark Michelson.</div>
<div>By rmudgett.</div>
<p style="color: grey;"><i>Updated 2011-04-26 17:49:41</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;">This patch fixes a couple connected line update problems:
1) The connected line needs to be updated when the initial INVITE is sent
if there is a peer callerid configured. Previously, the connected line
information did not get reported until the call was connected so SIP could
not report connected line information in ringing or progress messages.
2) The connected line should not be updated on initial connect if there is
no connected line information. Previously, all it did was wipe out any
default preset CONNECTEDLINE information set by the dialplan with empty
strings.
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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;">Both issues now work as intended.
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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=18367">18367</a>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/1.8/channels/chan_sip.c <span style="color: grey">(315663)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1199/diff/" style="margin-left: 3em;">View Diff</a></p>
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