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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3442/">https://reviewboard.asterisk.org/r/3442/</a>
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<p style="margin-top: 0;">On April 12th, 2014, 1:21 a.m. CDT, <b>Olle E Johansson</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/3442/diff/1/?file=57272#file57272line8262" style="color: black; font-weight: bold; text-decoration: underline;">/trunk/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 struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *title, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, struct ast_callid *callid)</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>pbx_builtin_setvar_helper(tmp, "TELPHONECONTEXT", i->tel_phone_context);</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;">Sadly, I think we should name this SIPURIPHONECONTEXT or something that starts with SIP like SIPDOMAIN SIPFROMADDRESS et al. Vanity tags may be a good thing.</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;">Works for me - changed.</pre>
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<p>- Matt</p>
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<p>On April 11th, 2014, 9:35 p.m. CDT, Matt Jordan wrote:</p>
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<div>Review request for Asterisk Developers, Geert Van Pamel and Olle E Johansson.</div>
<div>By Matt Jordan.</div>
<p style="color: grey;"><i>Updated April 11, 2014, 9:35 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-17179">ASTERISK-17179</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;">This patch is a continuation of 3349 (https://reviewboard.asterisk.org/r/3349/)
It resolves a finding oej had that the phone-context be available in a TELPHONECONTEXT channel variable. This patch adds that variable. It also allows a local number (or global number specified in the TEL URI) to be used to look up as a peer. Granted; not super useful. But hey.</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;">See review https://reviewboard.asterisk.org/r/3441</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/include/sip.h <span style="color: grey">(412292)</span></li>
<li>/trunk/channels/chan_sip.c <span style="color: grey">(412292)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3442/diff/" style="margin-left: 3em;">View Diff</a></p>
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