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
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
