Any reason why I don't get audio on the channel after it rings and the end user picks up. Here are my files.
CONSOLE=Console/dsp ; Console interface for demo OUTBOUNDTRUNK=SIP/callwithus [default] include => stdexten exten => s,1,Answer() exten => s,n,Wait(1) exten => s,n,Dial(SIP/callwithus/1111444444,120,A,(demo-thanks)) exten => s,n,Wait(2) exten => s,n,Hangup() my sip.conf file [general] context=default allowoverlap=no bindport=5060 port=5060 bindaddr=0.0.0.0 canreinvite=no ;if your asterisk box is behind a NAT ro ;register => xxx:y...@carrier.callwithus.com register => xxx:y...@sip.callwithus.com [callwithus] type=friend host=sip.callwithus.com username=xxx secret=yyy qualify=no insecure=invite -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
