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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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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">lmadsen: jrose: what are the chances you could make the option name something like: legacy_useroption_parsing={yes,no} or something like that?</pre>
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<p>- jrose</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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