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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3781/">https://reviewboard.asterisk.org/r/3781/</a>
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<p>Ship it!</p>
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<a href="https://reviewboard.asterisk.org/r/3781/diff/1/?file=63311#file63311line80" style="color: black; font-weight: bold; text-decoration: underline;">/trunk/channels/sip/dialplan_functions.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; ">int sip_acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "></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_copy_string</span><span class="p">(</span><span class="n">buf</span><span class="p">,</span> <span class="n">ast_sockaddr_isnull</span><span class="p">(</span><span class="o">&</span><span class="n">p</span><span class="o">-></span><span class="n">recv</span><span class="p">)</span> <span class="o">?</span> <span class="s">""</span> <span class="o">:</span> <span class="n">ast_sockaddr_stringify_fmt</span><span class="p">(</span><span class="o">&</span><span class="n">p</span><span class="o">-></span><span class="n">recv</span><span class="p">,</span> <span class="n">AST_SOCKADDR_STR_PORT</span><span class="p">),</span> <span class="n">buflen</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;">There's an inline function in include/asterisk/netsock2.h called ast_sockaddr_stringify_port() that you can use in place of the ast_sockaddr_stringify_fmt() call.</pre>
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<p>- Mark Michelson</p>
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<p>On July 21st, 2014, 1:15 p.m. UTC, dtryba wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By dtryba.</div>
<p style="color: grey;"><i>Updated July 21, 2014, 1:15 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-24040">ASTERISK-24040</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;">Retrieve the source port of an incoming (chan_sip) SIP invite in the dialplan with ${CHANNEL(recvport)}
To complement ${CHANNEL(recvip)} and enable me to make dialplan decisions based on source port (in a peerless setup, handle everything as guests using AGI to lookup source ip/port for routing/handling).
pjsip appears to have this capability through the CHANNEL function (pjsip,local_addr/remote_addr).
Simple 2 line patch using ast_sockaddr_stringify_fmt(const struct ast_sockaddr *sa, int format)
to return the port as a string.</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 on 11.10.2 (Debian Jessie) and trunk (418610) using IPv4. Having a few SIP endpoints connect from different address/ports combinations
Logged ${CHANNEL(recvip)}:${CHANNEL(recvport)} corresponds with source ip:port in packetdumps on the asterisk machine.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/trunk/channels/sip/dialplan_functions.c <span style="color: grey">(418610)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3781/diff/" style="margin-left: 3em;">View Diff</a></p>
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