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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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<a href="https://reviewboard.asterisk.org/r/3474/diff/1-2/?file=57790#file57790line21285" style="color: black; font-weight: bold; text-decoration: underline;">/branches/11/channels/chan_sip.c</a>
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)</pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span>/* add TLS and SRTP status of the channel */</pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span>/* add transport and media types */</pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span>ast_cli(a->fd, " Signaling: %s\n", cur->socket.type == SIP_TRANSPORT_TLS ? "TLS" : "non-TLS");</pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span>char *transport_type;</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;">Guidelines say you cannot declare a variable in the middle of a code block.</pre>
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<a href="https://reviewboard.asterisk.org/r/3474/diff/1-2/?file=57790#file57790line21287" style="color: black; font-weight: bold; text-decoration: underline;">/branches/11/channels/chan_sip.c</a>
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)</pre></td>
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<td bgcolor="#fdfebc" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span>ast_cli(a->fd, " Media: %s\n", cur->srtp ? "SRTP" : "RTP");</pre></td>
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<td bgcolor="#fdfebc" 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>} 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>} else if (cur->socket.type == SIP_TRANSPORT_WSS) {</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;">It would be better if this were put into its own function. Something like transport2str() that returns the transport string.</pre>
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<a href="https://reviewboard.asterisk.org/r/3474/diff/1-2/?file=57790#file57790line21297" style="color: black; font-weight: bold; text-decoration: underline;">/branches/11/channels/chan_sip.c</a>
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)</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;">Add curlies.</pre>
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<a href="https://reviewboard.asterisk.org/r/3474/diff/1-2/?file=57790#file57790line21299" style="color: black; font-weight: bold; text-decoration: underline;">/branches/11/channels/chan_sip.c</a>
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)</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;">red blob</pre>
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<p>- rmudgett</p>
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<p>On April 24th, 2014, 12:33 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 24, 2014, 12:33 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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