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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3474/">https://reviewboard.asterisk.org/r/3474/</a>
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<p style="margin-top: 0;">On April 25th, 2014, 1:03 p.m. CDT, <b>rmudgett</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/3474/diff/3/?file=57909#file57909line21287" style="color: black; font-weight: bold; text-decoration: underline;">/branches/11/channels/chan_sip.c</a>
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(Diff revision 3)
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static char *complete_sip_notify(const char *line, const char *word, int pos, int state)</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>if (cur->socket.type == SIP_TRANSPORT_TLS) {</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="tb"> </span>transport_type = "TLS";</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>} else if (cur->socket.type == SIP_TRANSPORT_UDP) {</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="tb"> </span>transport_type = "UDP";</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>} else if (cur->socket.type == SIP_TRANSPORT_TCP) {</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>} else if (cur->socket.type == SIP_TRANSPORT_WS) {</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="tb"> </span>transport_type = "WS";</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>} else if (cur->socket.type == SIP_TRANSPORT_WSS) {</pre></td>
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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;">These are supposed to be AST_TRANSPORT_xxx declarations. SIP_TRANSPORT_xxx declarations don't exist.
Please at least compile the patch.</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Heh. These were changed from SIP_TRANSPORT_xxx to AST_TRANSPORT_xxx in v12.</pre>
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<p>- rmudgett</p>
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<p>On April 25th, 2014, 12:37 p.m. CDT, Patrick Laimbock wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Patrick Laimbock.</div>
<p style="color: grey;"><i>Updated April 25, 2014, 12:37 p.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-23564">ASTERISK-23564</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;">AFAICT there is no way to see from the CLI if TLS and SRTP are enabled for a channel. I asked on the ML and in #asterisk but received no answer other than that nobody knew how to get that info from the CLI. This patch shows TLS or non-TLS and SRTP or RTP.</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;">Testing was done on Asterisk-11.8.1 with TLS & RPT, TLS & SRTP, non-TLS & RPT configured and a Nexus GSM using Linphone with similar configs. AFAICT the status of the channel and media was correctly reported for each scenario.</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/11/channels/chan_sip.c <span style="color: grey">(412921)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3474/diff/" style="margin-left: 3em;">View Diff</a></p>
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