<html>
<body>
<div style="font-family: Verdana, Arial, Helvetica, Sans-Serif;">
<table bgcolor="#f9f3c9" width="100%" cellpadding="8" style="border: 1px #c9c399 solid;">
<tr>
<td>
This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1886/">https://reviewboard.asterisk.org/r/1886/</a>
</td>
</tr>
</table>
<br />
<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Hey, would it be alright if I went ahead and submitted everything in this patch that isn't dahdi/iax2 related? The lack of reviewers for these areas is sort of holding things up for me.</pre>
<br />
<p>- jrose</p>
<br />
<p>On April 27th, 2012, 1:42 p.m., jrose wrote:</p>
<table bgcolor="#fefadf" width="100%" cellspacing="0" cellpadding="8" style="background-image: url('https://reviewboard.asterisk.org/media/rb/images/review_request_box_top_bg.png'); background-position: left top; background-repeat: repeat-x; border: 1px black solid;">
<tr>
<td>
<div>Review request for Asterisk Developers, Mark Michelson, rmudgett, and Matt Jordan.</div>
<div>By jrose.</div>
<p style="color: grey;"><i>Updated April 27, 2012, 1:42 p.m.</i></p>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" style="border: 1px solid #b8b5a0">
<tr>
<td>
<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;">https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging
The Unique Call ID logging patch progresses to its third phase. In this patch, I'm concentrating on moving the concept from something that is purely in pbx threads to something that lives in the channels themselves as well.
Covered in this rendition are chan_sip, chan_dahdi, and chan_iax2. DAHDI is somewhat unique in that it specifically targets the creation of new channels rather than through checking for new calls and requests.</pre>
</td>
</tr>
</table>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" style="border: 1px solid #b8b5a0">
<tr>
<td>
<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;">Testing has mostly involved the making of calls between various technologies and transferring. It'd be difficult to detail all of it, but some examples would include calls via dynamic pri spans in a loopback scenario, pots phones, SIP and IAX endpoints. So far most stuff is appearing about like expected. That is to say:
1. New calls get a callid an the channel sticks with it. Log messages between the sandwiched binding and unbinding of callid logging get logged with the apporpriate callids.
2. Transfers go one of two ways. In the case of blind transfers, the original callid is usually preserved while attended transfers result in a new call (which means a new callid) and the new callid becomes dominant for the call.
3. There is an exception to the above rule with transferring of calls that weren't bridged between two endpoints (IE, a call to a menu or some other asterisk pbx extension without dial). In this case, the callid that started the pbx thread is the one that ends up being used. Transferring these types of calls is a bit of a fringe case, but it's worth noting.
Any kind of bridging that involves a masqeurade is usually going to result in a new callid since fresh channels are being created for one end and/or another.</pre>
</td>
</tr>
</table>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
<ul style="margin-left: 3em; padding-left: 0;">
<li>/team/jrose/call_identifiers/channels/chan_dahdi.c <span style="color: grey">(363674)</span></li>
<li>/team/jrose/call_identifiers/channels/chan_iax2.c <span style="color: grey">(363674)</span></li>
<li>/team/jrose/call_identifiers/channels/chan_sip.c <span style="color: grey">(363674)</span></li>
<li>/team/jrose/call_identifiers/channels/sip/include/dialog.h <span style="color: grey">(363674)</span></li>
<li>/team/jrose/call_identifiers/channels/sip/include/sip.h <span style="color: grey">(363674)</span></li>
<li>/team/jrose/call_identifiers/include/asterisk/channel.h <span style="color: grey">(363674)</span></li>
<li>/team/jrose/call_identifiers/include/asterisk/logger.h <span style="color: grey">(363674)</span></li>
<li>/team/jrose/call_identifiers/main/channel.c <span style="color: grey">(363674)</span></li>
<li>/team/jrose/call_identifiers/main/channel_internal_api.c <span style="color: grey">(363674)</span></li>
<li>/team/jrose/call_identifiers/main/cli.c <span style="color: grey">(363674)</span></li>
<li>/team/jrose/call_identifiers/main/logger.c <span style="color: grey">(363674)</span></li>
<li>/team/jrose/call_identifiers/main/pbx.c <span style="color: grey">(363674)</span></li>
</ul>
<p><a href="https://reviewboard.asterisk.org/r/1886/diff/" style="margin-left: 3em;">View Diff</a></p>
</td>
</tr>
</table>
</div>
</body>
</html>