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VoIP Billing solution


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 Post subject: calling through provider ivr
PostPosted: Thu Aug 23, 2007 12:40 am 
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Joined: Fri Oct 27, 2006 6:17 pm
Posts: 161
i'need to make the call to one IVR, example in asterisk way:

exten => _00598.,1,Dial(SIP/PREFIX@PROVIDER,60,D(${EXTEN}))

I'need to do the same in a2b, remove the number dialled, and then when the ivr from my provider answer asking for the number send the number dialled from my client in DTMF mode.

is that posible?

Cheers.


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 Post subject:
PostPosted: Thu Aug 23, 2007 9:47 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
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


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 Post subject:
PostPosted: Fri Aug 24, 2007 4:17 pm 
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Joined: Fri Oct 27, 2006 6:17 pm
Posts: 161
already tried that and not work, and is the same than add the additional parameter in the trunk in a2b, i'need to remove the number dialled, and send then later the number in dtmf mode.


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 Post subject:
PostPosted: Fri Aug 24, 2007 7:19 pm 
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Joined: Mon Jun 19, 2006 5:14 pm
Posts: 218
why doesn't it work for you in the a2b trunk definition the example given is what you are looking for? no?

additional parameters:


Quote:
Define any additional parameters that will be used when running the Dial Command in Asterisk. Use the following tags as variables %dialingnumber%, %cardnumber%. ie 'D(ww%cardnumber%wwwwwwwwww%dialingnumber%)'


where the idea is to wait 1 sec (ww) send out the card number, wait 5 sec and then send the number to be dialed.

try something simple as this with your application. then when it works figure out how map ${exten} into this dial string


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 Post subject:
PostPosted: Sun Aug 26, 2007 7:56 pm 
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Joined: Fri Oct 27, 2006 6:17 pm
Posts: 161
yes that is working but i'dont know how remove the number dialled.

i'set in prefix the IVR number, but this is attached with the dialled number example :

SIP/provideIP/PREFIX+NUMBERDIALED

i'need to call

SIP/providerIP/prefix

and then when ask, send the number with 'D(%dialingnumber%)'

Cheers.


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VoIP Billing solution


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