[asterisk-users] Fax throughput - Asterisk 1.6.1.9

[Kevin P. Fleming]

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.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming at digium.com
Check us out at www.digium.com & www.asterisk.org