> Cyprus VoIP wrote: > >> Thank you for your answer. The 'internal extension' is indeed a T.38 >> capable device that works perfectly when connected directly to the >> Proxy/ITSP. >> >> As you said, the key to debugging/resolving this issue is the logger. I >> wasn't aware of this file. this is what I have there: >> ... >> ;debug => debug >> console => notice,warning,error >> ;console => notice,warning,error,debug >> messages => notice,warning,error >> ;full => notice,warning,error,debug,verbose >> ... >> >> Should I change the "console..." line or uncomment the ";full..." line? > > Either one is fine; using 'full' is actually a bit better, because the > color highlighting done on the console sometimes makes console captures > hard to read. >
Hi, So, I enabled the full logger, and the strange thing I see is this message: "Got T.38 Re-invite without audio. Keeping RTP active during T.38 session" It seems that this might be the reason Asterisk initiates a reINVITE with voice codecs, after connecting the 2 parties. Is there a way to disable that action, or do we need to add T.38 somehow to the list of codecs? I followed the instructions on the default sip.conf to include the line "t38pt_udptl=yes,redundancy" in the general section and in each of the parties. Thanks. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
