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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1235/">https://reviewboard.asterisk.org/r/1235/</a>
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<div>Review request for Asterisk Developers, Russell Bryant and David Vossel.</div>
<div>By jrose.</div>
<p style="color: grey;"><i>Updated 2011-05-31 11:22:36.724585</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;">Adds a specialized function similar to ast_encode_uri for display name that just escapes the quotation marks and doesn't do any other stuff like what's found in encode_uri.
Fixes a few stylistic concerns.
Tested escape_quotes by changing n to a variety of strings before invocation. Made sure it wouldn't segfault by overflowing the input buffer, but like before as when using ast_encode_uri if it cuts off early, it'll still leave the display name without an end quote and this will cause the uri to parse incorrectly and the call will fail.</pre>
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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;">Under SIP pedantic mode, this function was encoding strings meant for the display name into into a URI safe format when this was not specified in the SIP RFCs. I also took the liberty of adding a little commentary.
pedantic mode was the default in Asterisk 1.8+</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;">A number of different calls with user names including more interesting UTF-8 characters from a variety of characters from a variety of sources including trema (like German umlaut characters), Kanji (Japanese pictographic characters), Sanskrit (phonetic characters for one of India's languages), and Cyrillic (Russian). Whether or not they'll display properly depends on the receiving phone... My Grandstream phone doesn't like the higher end UTF-8 characters, but all of my soft phones read them fine.
Testing involved a manually set DAHDI channel to have odd caller ID and an incoming call from another Asterisk Box sending SIP from a phone set to also have a display name with high UTF-8 chars.</pre>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/view.php?id=18298">18298</a>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> (updated)</h1>
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<li>/branches/1.8/channels/chan_sip.c <span style="color: grey">(321510)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1235/diff/" style="margin-left: 3em;">View Diff</a></p>
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