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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2227/">https://reviewboard.asterisk.org/r/2227/</a>
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<div>Review request for Asterisk Developers.</div>
<div>By Olle E Johansson.</div>
<p style="color: grey;"><i>Updated April 11, 2014, 4:57 p.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;">Adding alert of new version.</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-20759">ASTERISK-20759</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 (updated)</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;">The SIP2CAUSE hangup code conversion tables has up to now been hard-coded in Asterisk. In some cases, like when building in-house ISDN/Q.SIG to SIP gateways, there's a need to manipulate this conversion.
With this code, advanced users can add a "private" conversion. This is added in front of the built-in conversions.
Asterisk conversion tables does not change in this patch. Everything should work as before. To shrink the chan_sip.c file a small bit I decided to move this functionality into a new source code file.
Adding:
- new source code file sip2cause.c and include file sip2cause.h
- new configuration file sip2cause.conf
Reviewboard doesn't seem accept the new files, so they have to be found in the branch itself.
http://svn.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-trunk
The new files are:
* http://svnview.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-trunk/configs/sip2cause.conf.sample
* http://svnview.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/sip2cause.c
* http://svnview.digium.com/svn/asterisk/team/oej/earl-grey-sip2cause-configurable-trunk/channels/sip/include/sip2cause.h
2014-04: A new version will be coming soon with a new function - custom hangupcauses outside of the ISDN range (as discussed on asterisk-dev a while ago).</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;">Tested all kinds of weird translations. This file should cause some errors (AST_CAUSE_SKREP doesn't exist, 903 is not a valid SIP reason code etc etc.
[sip2cause]
604 => AST_CAUSE_SKREP
404 => UNALLOCATED
599 Bad => USER_BUSY
486 => NORMAL_CLEARING
603 => UNALLOCATED
[cause2sip]
SKREP => 503 Service Failure
UNALLOCATED => 903 Go to hell
UNALLOCATED => 499 I don't want to do that.
USER_BUSY => 503 I am not feeling well</pre>
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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/include/sip_utils.h <span style="color: grey">(377205)</span></li>
<li>/trunk/channels/chan_sip.c <span style="color: grey">(377205)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2227/diff/" style="margin-left: 3em;">View Diff</a></p>
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