Hi
You could probably modify the dial command string in A2Billing.conf do do what you require: -
; More information about the Dial :
http://voip-info.org/wiki-Asterisk+cmd+dial
; 30 : The timeout parameter is optional. If not specifed, the Dial command will wait indefinitely, exiting only when the originating channel hangs up, or all the dialed channels return a busy or error condition. Otherwise it specifies a maximum time, in seconds, that the Dial command is to wait for a channel to answer.
; H: Allow the caller to hang up by dialing *
; r: Generate a ringing tone for the calling party
; R: Indicate ringing to the calling party when the called party indicates ringing, pass no audio until answered.
; m: Provide Music on Hold to the calling party until the called channel answers.
; L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms)
; %timeout% tag is replaced by the calculated timeout according the credit & destination rate!
dialcommand_param = "|60|HRrL(%timeout%:61000:30000)"
; by default (3600000 = 1HOUR MAX CALL)
dialcommand_param_sipiax_friend = "|60|HL(3600000:61000:30000)"
; Define the order to make the outbound call
; YES -> SIP/dialedphonenumber@gateway_ip - NO SIP/gateway_ip/dialedphonenumber
; Both should work exactly the same but i experimented one case when gateway was supporting dialedphonenumber@gateway_ip
; So in case of trouble, try it out
switchdialcommand = NO
Joe