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: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org _______________________________________________ -- 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
