Thank you Jonathan, I have read up on this, therefore 488 and the referenced 606 error, but I have to say it wouldn't have helped me. I find the description still very general. If one looks at the asterisk source code, then one can clearly find the case with the crypto line and missing RTP/SAVP. Describing these concrete occurences of errors is what I am currently missing.<br>
<br><div class="gmail_quote">2012/6/20 Jonathan Rose <span dir="ltr"><<a href="mailto:jrose@digium.com" target="_blank">jrose@digium.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="HOEnZb"><div class="h5">Stefan at WPF wrote:<br>
<br>
> is there anywhere an overview of SIP error codes and under which<br>
> condition they are reported by Asterisk?<br>
> There are general definitions for SIP error codes, but they are quite<br>
> general and it's Asterisk that actually checks what's wrong and then<br>
> reports an error. Now, currently I could check the source code to<br>
> get more informations what could have caused the error, but that's<br>
> very time consuming.<br>
><br>
> An example:<br>
> I recently had the "488 Not acceptable here" error. There were no<br>
> more details, only this error code. I had no idea what could cause<br>
> this error (what is not acceptable?) and where to start looking for<br>
> problems (except maybe check the source code of Asterisk). A<br>
> documentation of all possible SIP errors and under which conditions<br>
> they are reported - like the following example - would be very<br>
> helpful in such cases:<br>
><br>
> Description of "488 Not acceptable here"<br>
> - Could be caused by codec problems, when codec negotiation failed.<br>
> You can check if the negotiation failed by ....<br>
> - Can be paused by a phone offering encryption, but only offering<br>
> RTP/AVP instead of RTP/SAVP profile. Check if the sip log contains a<br>
> crypto line and only RTP/AVP, if yes, change the phone settings from<br>
> RTP/AVP to RTP/SAVP or disable RTP encryption in the phone's<br>
> settings.<br>
> [Even better: Besides throwing the error message also add the reason<br>
> for it, at least in the Asterisk log files. I had a warning from<br>
> Asterisk before the error code, but a warning is still something<br>
> different than an error, for me the relation between both, the<br>
> warning and the error message, weren't clear]<br>
><br>
><br>
> Is there something like this already? How about introducing it, e.g.<br>
> every Asterisk developer throwing an error message in his code adds<br>
> the reason for throwing the error message to an explanation of<br>
> possible causes, like in the example above?<br>
><br>
> Best regards<br>
> Stefan<br>
><br>
<br>
</div></div>Hi Stefan, it's hardly Asterisk specific, but I'd recommend you<br>
try RFC 3261 <a href="http://www.ietf.org/rfc/rfc3261.txt" target="_blank">http://www.ietf.org/rfc/rfc3261.txt</a><br>
<br>
In section 21.4, most if not all of the SIP 4XX request errors are<br>
mentioned including the one you just noted (488).<br>
<br>
--<br>
Jonathan R. Rose<br>
Digium, Inc. | Software Engineer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - US<br>
direct <a href="tel:%2B1%20256%20428%206139" value="+12564286139">+1 256 428 6139</a><br>
<br>
Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a><br>
<br>
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