On Fri, 2003-10-24 at 21:12, David J Carter wrote: > Thanks Dave, > > I can now call a sipphone number from * but get no voice throughput. > > I still don't see anything coming in from sipphone though. > > Dave >
Have a look at rtp.conf I have 8000 - 8060 there. by default its 10000, if you have a Grandstream you must tell it what to use. It works because sipphone newbies keep calling me. They can't believe transatlantic is so good. -- Dave Cotton <[EMAIL PROTECTED]> _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
