<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;"><br><div><div>On 29 Oct 2013, at 19:00, Scott Griepentrog <<a href="mailto:sgriepentrog@digium.com">sgriepentrog@digium.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><div dir="ltr"><div class="gmail_default" style="color:rgb(102,0,0)">I like that idea. A value >128 means a SIP code, and <128 means an ISDN code, and you can then use the original native code or translate to the desired range by using a to_sip() or to_isdn() function.</div>
<div class="gmail_default" style="color:rgb(102,0,0)"><br></div><div class="gmail_default" style="color:rgb(102,0,0)">Or would it create inconsistencies that would break customer's solutions?</div></div></blockquote><br>The problem is that we're still a multiprotocol PBX. Switching to SIP may hurt other channels</div><div>and be as stupid as ISDN. Now, SIP has more finegrained error codes than ISDN, so translation</div><div>will be better for most channels.</div><div><br></div><div>I know that I'm contradicting myself. That's an art form you learn as you grow older ;-)</div><div>/O<br><blockquote type="cite"><div class="gmail_extra">
<br><br><div class="gmail_quote">On Tue, Oct 29, 2013 at 11:21 AM, Tilghman Lesher <span dir="ltr"><<a href="mailto:tilghman@meg.abyt.es" target="_blank">tilghman@meg.abyt.es</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="im">On Tue, Oct 29, 2013 at 10:43 AM, Olle E. Johansson <<a href="mailto:oej@edvina.net">oej@edvina.net</a>> wrote:<br>
><br>
> On 29 Oct 2013, at 15:33, Tilghman Lesher <<a href="mailto:tilghman@meg.abyt.es">tilghman@meg.abyt.es</a>> wrote:<br>
><br>
>> On Tue, Oct 29, 2013 at 4:01 AM, Olle E. Johansson <<a href="mailto:oej@edvina.net">oej@edvina.net</a>> wrote:<br>
</div><div class="im">>>> The questions are:<br>
>>> - is this too dangerous, to grab some ISDN cause codes?<br>
>>> - will it have side effects?<br>
>>> - what will other channels do if they get a hangup(119) ?<br>
>>><br>
>>> All of this is of course a bit dangerous and have to be used only by<br>
>>> experienced asterisk admins, but is something that have been asked for for a<br>
>>> long time. There is nothing stopping you from redefining any cause code to<br>
>>> 401, 180 or something else, like 302. But that will hurt...<br>
>><br>
>> The only thing I fear from such a change is that it will lead to a<br>
>> support nightmare if any of the cause codes that you've appropriated<br>
>> are eventually used in another way by the ITU. Since that's the<br>
>> standards body involved with the definition of ISDN, I'd suggest that<br>
>> you obtain from them an assurance that such cause codes land in a<br>
>> "reserved" or "vendor-defined" state, to avoid this potential<br>
>> future-compatibility issue. It also creates a potential future issue<br>
>> within Asterisk, whereby if the lowest cause code is allocated, and<br>
>> that custom code needs to be moved, then there will be an issue with<br>
>> "custom1" being assigned to a different integer on two different<br>
>> versions of Asterisk. This only potentially matters when the cause<br>
>> code is transmitted externally, as we might in the ISDN driver.<br>
><br>
> Noted.<br>
><br>
> Eventually I want a core that is not based on ISDN anymore. But that's another story.<br>
<br>
</div>Speaking historically, we could just have easily chosen SIP cause<br>
codes as the internal representation. Or we could have made up our<br>
own. For that matter, if the issue is backwards compatibility to<br>
ISDN, then using values in the 128+ range would make sense, possibly<br>
enjoining the ISDN driver from sending those codes directly. Assuming<br>
that you wouldn't want to ever send a SIP cause code lower than 128,<br>
using the SIP cause code directly in this field (it is, after all, a<br>
full integer) may be preferable. If you would need to send lower<br>
values, however, then multiplying the SIP cause code by a certain<br>
factor (say 1,000) may be sufficient. Changes would need to be made,<br>
though, to ensure that other areas of Asterisk do not assume that the<br>
value will fit within a single 8-bit field.<br>
<span class="HOEnZb"><font color="#888888"><br>
-Tilghman<br>
</font></span><div class="HOEnZb"><div class="h5"><br>
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<strong style="font-family: Arial, Helvetica, sans-serif; font-size: 12px;">Scott Griepentrog</strong><br style="font-family: Arial, Helvetica, sans-serif; font-size: 12px;"><span style="font-family: Arial, Helvetica, sans-serif; font-size: 12px;">Digium, Inc · Software Developer</span><br style="font-family: Arial, Helvetica, sans-serif; font-size: 12px;">
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