Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through which I connected an external PSTN line. I use it as carrier for VoIP calls. I can make successfully calls, but there's one problem, I receive 200 OK with SDP with delay (sometimes more than 30 seconds). So when I make a call through asterisk I receive intially: - 100 Trying - 183 Session Progress, with SDP when the called number respond, I start receiving RTP with voice, also the called receives voice from me, but only after a while asterisk sends 200 OK with SDP.
I'm not sure if the problem is from asterisk or from the telephony provider (I think the provider). Is there a posibility to replace 183 with 200 OK ? I mean from the moment when ringing starts to receive 200 OK with SDP instead of 183 ? Thank you, Marius _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
