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Predictive Dialer


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 Post subject: strip DNID (access number) from called number
PostPosted: Mon Mar 11, 2013 7:57 pm 
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Joined: Mon Mar 11, 2013 7:23 pm
Posts: 9
hi great forum! :D

i am new to a2billing and i am trying to figure out the following suituation

i want my client to call a local acess DID (21xxxxxxxx)+ number he would like to call (31xxxxxxxxx) and without any voice promts to get immediate access to desired number (31xxxxxxxxx)

1. i have made custom destination for DID -> a2billing using conf1
2. i have well made rates tested with simulator (31)
3. i enabled clid auth and its working well
4. this is my Custom Destination: a2billing,${EXTEN},1
5. extensions_custom.conf :
[a2billing]

exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,DeadAGI(a2billing.php,1)
exten => _X.,n,Hangup


the problem i am facing is that when the client calls 21xxxxxxxx31xxxxxxxxx a2billing trying to match 21xxxxxxx
and obviously cant because the access number its not what i want to match and i havent included in rates too

i also tried to REMOVE_PREFIX_from_trank = 21xxxxxxxx wint no success

here is the a2billing log
Code:
WHERE cc_tariffgroup.id=2 AND ((dialprefix='21xxxxxxx' OR dialprefix='21xxxxxx' OR ..... uniqueid:1363027455.334]:[CallerID:21xxxxxx]:[CN:]:[ERROR ::> RateEngine didnt succeed to match the dialed numbe


and asterisk log
Code:
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/301-00000113", "SIP/trunk-voip/21xxxxxxxx31xxxxxxx,300,") in new stack------------- -- Called SIP/trunk-voip/21xxxxxxxx31xxxxxxxx -------------- -- Executing [21xxxxxxxxx@from-trunk:6] Goto("SIP/access-00000115", "a2billing,21xxxxxxx,1") in new stack
    -- Goto (a2billing,21xxxxxxxx,1)[quote]"NUMBER DIDNT SENT TO A2BILLING"[/quote]
-
\

as i can understand real number never leaves asterisk and pass to a2billing

thanks .


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 Post subject: Re: strip DNID (access number) from called number
PostPosted: Tue Mar 12, 2013 10:23 am 
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Joined: Mon Mar 02, 2009 8:56 pm
Posts: 271
Where is the customer dialling the number from?

You say they are calling a local access DID but this shows the call going out from (an?) Asterisk system -

SIP/trunk-voip/21xxxxxxxx31xxxxxxx

Are they calling from a regulat PSTN phone, or from the A2Billing/Asterisk system itself?


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