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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1223/">https://reviewboard.asterisk.org/r/1223/</a>
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<p style="margin-top: 0;">On May 19th, 2011, 5:07 p.m., <b>David Vossel</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/1223/diff/1/?file=16574#file16574line435" style="color: black; font-weight: bold; text-decoration: underline;">/branches/1.8/configs/sip.conf.sample</a>
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(Diff revision 1)
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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; ">;legacyuseroptionparsing=yes Default "no"</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; "> If you have this option enabled and there are semicolons</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> in the user field of a sip URI, the field be truncated</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> at the first semicolon seen. This effectively makes</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> semicolon a non-usable character for peer names, extensions,</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> and maybe other, less tested things. This can be useful</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> for improving compatability with devices that like to use</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> user options for whatever reason. The behavior is similar to</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> how SIP URI's were typically handled in 1.6.2, hence the name.</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;">Use spaces on these lines instead of tabs.</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;">Yeah, and underscores in the option name like leif requested as well.</pre>
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<p>- David</p>
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<p>On May 19th, 2011, 2:02 p.m., jrose wrote:</p>
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<div>Review request for Asterisk Developers, Russell Bryant, David Vossel, and Leif Madsen.</div>
<div>By jrose.</div>
<p style="color: grey;"><i>Updated 2011-05-19 14:02:21</i></p>
<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;">A quick recap:
As of Asterisk 1.8, semicolons in user fields are accepted as just part of the user field in compliance with RFC 3261. This makes devices that employ those options unable to register and also unable to match intended extensions without dialplan workarounds.
This approach involves a global sip option (if it were done per channel, we couldn't match on registers unfortunately) to strip the semicolons in the same general way as it was done in 1.6.2. For that reason I chose to call it legacyuseroptionparsing</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;">Made sure matches while the option were on would happen with the following using sipp:
REGISTER sip:localhost SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
To: "Jonathan" <sip:evilhost;garbage@[local_ip]:[local_port]>
From: "Jonathan" <sip:evilhost;garbage@[local_ip]:[local_port]>;tag=[call_number]
Call-ID: [call_id]
CSeq: 1 REGISTER
Contact: sip:evilhost@[local_ip]:[local_port];expires=3600
ALLOW: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Sipp
Content-Length: [len]
And
INVITE sip:2005;5002;phone-context=+1;npdi=yes@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "Lrrrr Schmrrr" <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: Asterisk <sip:2005;5002;phone-context=+1;npdi=yes@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 OPTIONS
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Asterisk Testsuite
Content-Length: [len]
Also that they also acted the same as the way they acted before with the option off.</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=18344">18344</a>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/1.8/channels/chan_sip.c <span style="color: grey">(319526)</span></li>
<li>/branches/1.8/channels/sip/include/sip.h <span style="color: grey">(319526)</span></li>
<li>/branches/1.8/configs/sip.conf.sample <span style="color: grey">(319526)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1223/diff/" style="margin-left: 3em;">View Diff</a></p>
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