[asterisk-users] Overview of SIP error codes and possible causes?

[Jonathan Rose]

Stefan at WPF wrote:

> is there anywhere an overview of SIP error codes and under which
> condition they are reported by Asterisk?
> There are general definitions for SIP error codes, but they are quite
> general and it's Asterisk that actually checks what's wrong and then
> reports an error. Now, currently I could check the source code to
> get more informations what could have caused the error, but that's
> very time consuming.
> 
> An example:
> I recently had the "488 Not acceptable here" error. There were no
> more details, only this error code. I had no idea what could cause
> this error (what is not acceptable?) and where to start looking for
> problems (except maybe check the source code of Asterisk). A
> documentation of all possible SIP errors and under which conditions
> they are reported - like the following example - would be very
> helpful in such cases:
> 
> Description of "488 Not acceptable here"
> - Could be caused by codec problems, when codec negotiation failed.
> You can check if the negotiation failed by ....
> - Can be paused by a phone offering encryption, but only offering
> RTP/AVP instead of RTP/SAVP profile. Check if the sip log contains a
> crypto line and only RTP/AVP, if yes, change the phone settings from
> RTP/AVP to RTP/SAVP or disable RTP encryption in the phone's
> settings.
> [Even better: Besides throwing the error message also add the reason
> for it, at least in the Asterisk log files. I had a warning from
> Asterisk before the error code, but a warning is still something
> different than an error, for me the relation between both, the
> warning and the error message, weren't clear]
> 
> 
> Is there something like this already? How about introducing it, e.g.
> every Asterisk developer throwing an error message in his code adds
> the reason for throwing the error message to an explanation of
> possible causes, like in the example above?
> 
> Best regards
> Stefan
> 

Hi Stefan, it's hardly Asterisk specific, but I'd recommend you
try RFC 3261 http://www.ietf.org/rfc/rfc3261.txt

In section 21.4, most if not all of the SIP 4XX request errors are
mentioned including the one you just noted (488).

--
Jonathan R. Rose
Digium, Inc. | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct +1 256 428 6139 

Check us out at: http://digium.com & http://asterisk.org