[asterisk-users] Fax throughput - Asterisk 1.6.1.9

Cyprus VoIP voipcy at gmail.com
Fri Dec 4 03:26:50 CST 2009


> 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.



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