Hi Bill, Your problem seems to be a codec negotiation issue, I think you need to specify for each SIP peer: disallow=all allow=alaw allow=ulaw ; and any others that you might need
Paul > ----- Original Message ----- > From: Bill Michaelson > To: [EMAIL PROTECTED] > Sent: Tuesday, February 10, 2004 9:26 AM > Subject: Re: [Asterisk-Users] asterisk-grandstream call > > > Arg.. my posting was mangled by text-wrapping. Sorry. > > Here again... > sip.conf: > > [general] > port = 5060 ; Port to bind to > bindaddr = 0.0.0.0 ; Address to bind to > context = default ; Default for incoming calls > [248379] > username=billdesk > type=friend > host=dynamic > canreinvite=no > mailbox=1234 > context=demo _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
