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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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<p style="margin-top: 0;">On March 22nd, 2014, 4:39 p.m. CET, <b>Olle E Johansson</b> wrote:</p>
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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;">I don't see what happens with the phone-context argument. Shouldn't we pass that on as a channel variable? </pre>
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<p>On March 22nd, 2014, 9:23 p.m. CET, <b>Geert Van Pamel</b> wrote:</p>
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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;">We return this into the hostport.</pre>
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<p>On March 22nd, 2014, 10:01 p.m. CET, <b>Geert Van Pamel</b> wrote:</p>
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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;">According to RFC 3966 phone-context is either a domain-name, or (part of) an international telephone number (indicated with +prefix).
It is used by a gateway to know how to dial the "local" number... the local number must be unique within its context...</pre>
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<p>On March 23rd, 2014, 6 p.m. CET, <b>Olle E Johansson</b> wrote:</p>
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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;">So it ends up in the SIPDOMAIN variable in the dial plan? It has to be reachable in the dial plan somehow. </pre>
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<p>On March 23rd, 2014, 10:12 p.m. CET, <b>Geert Van Pamel</b> wrote:</p>
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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;">The variable ${SIPDOMAIN} contains the local IP address of the Asterisk server.
The userinfo arrives in ${CALLERID} and is displayed on the display of the called device, and arrives in the CDR file.
Actually I do not know into which variable the incoming hostport info is copied to?
Could somebody else answer this question?</pre>
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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;">If I place a normal call to sip:geert@example.com to my Asterisk server. "geert" will be the extension I'm looking for, "example.com" ends up in SIPDOMAIN. It's not the local IP address, it's the domain/host part of the request URI in the INVITE.
I would prefer if phone context ended up in TELPHONECONTEXT so I could use it the same way as SIPDOMAIN in the dial plan. It should not end up in SIPDOMAIN as it is not a SIP uri. That way an extension in a local context can be routed differently than an extension in a global context.</pre>
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<p>- Olle E</p>
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<p>On March 22nd, 2014, 2:08 p.m. CET, Geert Van Pamel wrote:</p>
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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 22, 2014, 2:08 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;">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.
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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/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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