asterisk users - Jun 2012 - Overview of SIP error codes and possible causes?

Hello,

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
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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