You are right, I think the provider has some problems, and this should be fixed. But is also good to know that I can do this workaround for the worse case scenario.
thank you ! Olle E. Johansson wrote: > 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 > > _______________________________________________ -- 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
