; enable the option to call sip/iax friend for free (values : YES - NO)
; if SIP_IAX_FRIENDS is active, you can define a prefix for the dialed digits to call a pstn number
; values : number
; this will enable a prompt to enter your destination number.
; if number start by sip_iax_pstn_direct_call_prefix we do directly a sip iax call, if not we do a normal call
; Extracharge DIDs, multiple numbers and fees must be separated by comma
; 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
; 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!
; by default (3600000 = 1HOUR MAX CALL)
; 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
; For free calls, limit the duration: amount in seconds
maxtime_tocall_negatif_free_route = 5400