[asterisk-users] Fax throughput - Asterisk 1.6.1.9

[Cyprus VoIP]

> Cyprus VoIP wrote:
> 
>> 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.
> 
> Sorry, that's not the issue. That just means that chan_sip didn't
> destroy the internal RTP structures used for the audio part of the call
> when the call switched to T.38, which is only an optimization so we
> don't have to recreate them if the call switches back.
> 

Hi Kevin,

Thank you for your support.

If it's not related, why does Asterisk send again INVITE messages to 
both parties? How can this be prevented? I don't see more debug data 
prior to the new INVITE.

Thanks.