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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
