[Asterisk-Users] SIP debugging

[Klaus Darilion]

Kevin P. Fleming wrote:
> Klaus Darilion wrote:
> 
>> I've problems with Asterisk 1.2(svn trunk). Sometimes Asterisk does not
>> accept the 200 OK responses. E.g in the following example, Asterisk
>> retransmits the CANCEL although the 200 OK is received.
> 
> SVN trunk is not Asterisk 1.2.

Of course - sorry. I've meant Asterisk 1.2 from SVN branch 1.2
> 
> There is no way to help you with this partial SIP trace, and without any
> Asterisk version or configuration information. Asking 'smart questions'
> usually leads to people being able to help you :-)

IMO this was a smart question. I did not asked to debug my call flows, 
but I asked how can I debug it myself. For some reason Asterisk does not 
like my SIP responses, but there is no Warning, Error or any other log 
message although verbose=9 and "sip debug".

Shouldn't there be some error indication if Asterisk discards a response?

thanks
Klaus