U need 2 give more info on your setup i.e wherether u have sipclient<<>>asterisk<<>>nat<<>>sipclient or whatever the situation is . Anyway in the mean time just rtp dubug on and see wherether there r rtp packets sent back and forth
> > there's actually no audio b/w sip to sip calls. > > I just tried 2 sip extensions and there was no audio in any of them. > > what could be wrong? > > nat = 1 > > i used ulaw and then gsm and one of them worked. > > im not using qualify. i have tried everything i could think of, even > applied the patch < mydiff.txt in the src directory and did the make > clean; make install. > > John > > Quoting [EMAIL PROTECTED]: > >> Hi im experiencing no audio problems. ive installed the latest >> asterisk 1.2.10 zaptel, libpri & asterisk. >> >> the caller's side reception is fine but i hear nothing on my sip >> account. >> >> Please help >> Regards, >> John > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users