Un-top-posting... >> 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))
> On Sun, Dec 5, 2010 at 8:34 PM, Steve Edwards <[email protected]> > wrote: > >> Re-read 'core show application dial' >> >> Where is your prompt to option 'A' ? On Sun, 5 Dec 2010, Thomas Perron wrote: > negative. no joy. > removed the line to make is very basic. see below. > exten => s,1,Answer() > exten => s,n,Wait(1) > exten => s,n,Dial(SIP/callwithus/1111444444) Crank up the verbosity and debugging levels, check the codecs, etc. Does 'sip set debug on' give any clues? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards [email protected] 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
