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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1504/">https://reviewboard.asterisk.org/r/1504/</a>
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<p>Ship it!</p>
<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">The code looks good. I don't know if it is standards compliant or not, but I guess if there are devices that expect this behavior and it fixes a regression, we should make this change.</pre>
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<a href="https://reviewboard.asterisk.org/r/1504/diff/1/?file=21050#file21050line8497" 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 get_ip_and_port_from_sdp(struct sip_request *req, const enum media_type media, struct ast_sockaddr *addr)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="cm">/*! \brief If the address is null or ANY / unspecified (0.0.0.0 or ::), returns true; false otherwise.</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="cm"> * In some circumstances, calls should be placed on hold if either of these conditions exist.</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="cm"> */</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;">Use \retval in the docs. Also tag this sd \internal.</pre>
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<p>- Matthew</p>
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<p>On October 10th, 2011, 12:53 p.m., mjordan wrote:</p>
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<div>Review request for Asterisk Developers and Tzafrir Cohen.</div>
<div>By mjordan.</div>
<p style="color: grey;"><i>Updated Oct. 10, 2011, 12:53 p.m.</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;">As reported, Asterisk (1.6.2 and prior) would previously place calls on hold if an INVITE was received with c=0.0.0.0. This fails in 1.8 and later, as we only check to see if the various addresses are null.
This patch resolves this by checking if the addresses received are null or any. This is done using the netsock2 library of functions, so it should be compatible with both IPv4 and IPv6.</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;">Tested using a cobbled together SIPp scenario, which appeared to behave as expected. Tested also under the Asterisk TestSuite, which continued to work as expected.</pre>
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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-18086">ASTERISK-18086</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">(340135)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1504/diff/" style="margin-left: 3em;">View Diff</a></p>
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