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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3349/">https://reviewboard.asterisk.org/r/3349/</a>
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<div>Review request for Asterisk Developers, Corey Farrell, lmadensen, Matt Jordan, and wdoekes.</div>
<div>By Geert Van Pamel.</div>
<p style="color: grey;"><i>Updated March 19, 2014, 10:01 a.m.</i></p>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Changes</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;">parse_uri_full: do not quote RFC 3966 - only referring;
corrected multi-line comment format.
AST_TEST_DEFINE:
Corrected sip_parse_uri_full_test (added missing tel: schema).
Corrected sip_parse_uri_test: testing results.</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-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;">Implements RFC-3966 TEL URI incoming INVITE.
See https://issues.asterisk.org/jira/browse/ASTERISK-17179 for a description of the original isssue.
I have been patching all versions since Asterisk 1.6. I would like to include the code into the main trunk for version 13.
Previously Asterisk was failing with error on incoming IMS call:
Nov 13 17:52:05 NOTICE[27459]: chan_sip.c:6973 check_user_full: From address missing 'sip:', using it anyway
Nov 13 17:52:05 WARNING[27459]: chan_sip.c:6525 get_destination: Huh? Not a SIP header (tel:0987654321;phone-context=+32987654321)?
Reason: tel: protocol was not recognized.</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;">Executed an incoming TEL URI INVITE connection.
CLI was present on the display and in the CDR file.
No errors on SIP debug output.
</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> (updated)</h1>
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<li>/trunk/channels/sip/reqresp_parser.c <span style="color: grey">(410429)</span></li>
<li>/trunk/channels/chan_sip.c <span style="color: grey">(410429)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3349/diff/" style="margin-left: 3em;">View Diff</a></p>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">File Attachments </h1>
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<li><a href="https://reviewboard.asterisk.org/media/uploaded/files/2014/03/13/cad7a996-88c1-47fe-a2a9-cc6987af3b75__rfc-3966-tel-uri-patch-diff.txt">RFC-3966 tel URI patch</a></li>
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