> 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. _______________________________________________ -- 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
