Steve, thanks for your note negative. no joy. removed the line to make is very basic. see below.
[globals] CONSOLE=Console/dsp ; Console interface for demo OUTBOUNDTRUNK=SIP/callwithus ;[general] [default] include => stdexten exten => s,1,Answer() exten => s,n,Wait(1) exten => s,n,Dial(SIP/callwithus/1111444444) exten => s,n,Wait(2) exten => s,n,Hangup() ~ On Sun, Dec 5, 2010 at 8:34 PM, Steve Edwards <asterisk....@sedwards.com> wrote: > On Sun, 5 Dec 2010, Thomas Perron wrote: > >> Any reason why I don't get audio on the channel after it rings and the >> end user picks up. > >> exten => s,n,Dial(SIP/callwithus/1111444444,120,A,(demo-thanks)) > > Re-read 'core show application dial' > > Where is your prompt to option 'A' ? > > -- > Thanks in advance, > ------------------------------------------------------------------------- > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _____________________________________________________________________ > -- 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 > -- _____________________________________________________________________ -- 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