4 sep 2009 kl. 13.40 skrev Marius Ciorecan: > 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 ? >
You can answer() at any point in the dialplan - and that will generate a 200 OK. Like exten => marius,1,answer() exten => marius,n,dial(sip/mariusphone) This will generate an immediate 200 ok, regardless if mariusphone is busy or gone from the network. It's propably not what you want. Asterisk sends 200 OK on the incoming call as soon as we get a connection reply, a 200 OK or something similar in other protocols on the outbound call. For some reason, this happens very late for you and causes your problem. Could be some issue with the service provider, your ISDN connection or -even worse - your IAX2 trunk... (could not resist) Please start with debugging that and solving the real issue, instead of trying to change the functionality in Asterisk :-) Regards, /O --- o...@edvina.net - http://edvina.net Open Unified Communication - building platforms with SIP and XMPP From PBX to large scale implementations for carriers. Contact us today! _______________________________________________ -- 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