[asterisk-users] a2billing for residential voip usage

Vardan Harutyunyan hvardan71 at gmail.com
Tue Jun 15 04:43:27 CDT 2010


And also, what a2b version you are use?

If you are use 1.7 then all config is in DB, if 1.3(4) all config in 
a2billing.conf



-- 
Vardan Harutyunyan,
Senior System Administrator

Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: info at eif.am
www.eif-it.com

Vardan Harutyunyan wrote:
> I send you my a2b config for whole sale
>
> use_dnid = YES - this is the main option that you must use
>
> You can call this config like so:
> DeadAGI(a2billing.php|3)
>
> I hope this will be help you.
>
> [agi-conf3]
>
> ; the debug level
> ; 0=none, 1=low, 2=normal, 3=all
> debug = 0
>
> ; Asterisk Version Information
> ; 1_1,1_2,1_4 By Default it will take 1_2 or higher
> asterisk_version = 1_4
>
> ; Manage the answer on the call
> answer_call = NO
>
> ; Play audio - this will disable all stream file but not the Get Data
> ; for wholesale ensure that the authentication works and than number_try = 1
> play_audio = NO
>
> ; play the goodbye message when the user has finished.
> say_goodbye = NO
>
> ; enable the menu to choose the language
> ; press 1 for English, pulsa 2 para el espaУБol, Pressez 3 pour FranУЇais
> play_menulanguage = NO
>
>
> ; force the use of a language, if you dont want to use it leave the
> option empty
> ; Values : ES, EN, FR, etc... (according to the audio you have installed)
> force_language =
>
> ; Introduction prompt : to specify an additional prompt to play at the
> beginning of the application
> intro_prompt =
>
> ; Minimum amount of credit to use the application
> min_credit_2call = 0
>
> ; this is the minimum duration in seconds of a call in order to be billed
> ; any call with a length less than min_duration_2bill will have a 0 cost
> ; useful not to charge callers for system errors when a call was
> answered but it actually didn't connect
> min_duration_2bill = 0
>
> ; if user doesn't have enough credit to call a destination, prompt him
> to enter another cardnumber
> notenoughcredit_cardnumber = NO
>
> ; if notenoughcredit_cardnumber = YES  then     assign the CallerID to
> the new cardnumber
> notenoughcredit_assign_newcardnumber_cid = NO
>
>
> ; if YES it will use the DNID and try to dial out, without asking for
> the phonenumber to call
> ; value : YES, NO
> use_dnid = YES
>
> ; list the dnid on which you want to avoid the use of the previous
> option "use_dnid"
> no_auth_dnid = 2400,2300
>
> ; number of times the user can dial different number
> number_try = 1
>
> ; this will force to select a specific call plan by the Rate Engine
> force_callplan_id  =
>
> ; Play the balance to the user after the authentication (values : yes - no)
> say_balance_after_auth = NO
>
> ; Play the balance to the user after the call (values : yes - no)
> say_balance_after_call = NO
>
> ; Play the initial cost of the route (values : yes - no)
> say_rateinitial = NO
>
> ; Play the amount of time that the user can call (values : yes - no)
> say_timetocall = NO
>
>
> ; enable the setup of the callerID number before the outbound is made,
> by default the user callerID value will be use
> auto_setcallerid = NO
>
> ; If auto_setcallerid is enabled, the value of force_callerid will be
> set as CallerID
> force_callerid =
>
> ; If force_callerid is not set, then the following option ensures that
> CID is set to one of the card's configured caller IDs or blank if none
> available.
> ; NO - disable this feature, caller ID can be anything.
> ; CID - Caller ID must be one of the customers caller IDs
> ; DID - Caller ID must be one of the customers DID nos.
> ; BOTH - Caller ID must be one of the above two items.
> cid_sanitize = NO
>
>
> ; enable the callerid authentication
> ; if this option is active the CC system will check the CID of caller
> cid_enable = NO
>
> ; if the CID does not exist, then the caller will be prompt to enter his
> cardnumber
> cid_askpincode_ifnot_callerid = NO
>
> ; if the callerID authentication is enable and the authentication fails
> then the user will be prompt to enter his cardnumber
> ; this option will bound the cardnumber entered to the current callerID
> so that next call will be directly authenticate
> cid_auto_assign_card_to_cid = NO
>
> ; if the callerID is captured on a2billing, this option will create
> automatically a new card and add the callerID to it
> cid_auto_create_card = NO
>
> ; set the length of the card that will be auto create (ie, 10)
> cid_auto_create_card_len = 10
>
> ; If cid_auto_create_card has been set to YES, the following options
> will define with which configuration we will create the card
> ;
> ; billing type of the new card
> ; ( value : POSTPAY or PREPAY)
> cid_auto_create_card_typepaid = POSTPAY
>
> ; amount of credit of the new card
> cid_auto_create_card_credit = 0
>
> ; if postpay, define the credit limit for the card
> cid_auto_create_card_credit_limit = 1000
>
> ; the tariffgroup to use for the new card (this is the ID that you can
> find on the admin web interface)
> cid_auto_create_card_tariffgroup = 6
>
> ; to check callerID over the cardnumber authentication (to guard against
> spoofing)
> callerid_authentication_over_cardnumber = NO
>
> ; enable the option to call sip/iax friend for free (values : YES - NO)
> sip_iax_friends = no
>
> ; if SIP_IAX_FRIENDS is active, you can define a prefix for the dialed
> digits to call a pstn number
> ; values : number
> sip_iax_pstn_direct_call_prefix = 555
>
> ; 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
> sip_iax_pstn_direct_call = NO
>
> ; enable the option to refill card with voucher in IVR (values : YES - NO)
> ivr_voucher = NO
>
> ; if ivr_voucher is active, you can define a prefix for the voucher
> number to refill your card
> ; values : number - don't forget to change
> prepaid-refill_card_with_voucher audio accordingly
> ivr_voucher_prefix = 8
>
> ; When the user credit are below the minimum credit to call min_credit
> ; jump directly to the voucher IVR menu  (values: YES - NO)
> jump_voucher_if_min_credit = NO
>
> ; Extracharge DIDs, multiple numbers and fees must be separated by comma
> ; extracharge_did = 1800XXXXXXX,1888XXXXXXX
> extracharge_did =
> ;extracharge_fee = 0.02,0.03
> extracharge_fee =
>
> ; List the prefixes that will be stripped off if the call plan requires it
> international_prefixes = 9999999999999
>
> ; 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
> ;       g: When the called party hangs up, exit to execute more commands
> in the current context. (new in 1.4)
> ;       i: Asterisk will ignore any forwarding (302 Redirect) requests
> received. Essential for DID usage to prevent fraud. (new in 1.4) Useful
> if you are ringing a group of people and one person has set their phone
> to forwarded direct to voicemail on their cell or something which
> normally prevents any of the other phones from ringing.
> ;       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|HRgrL(%timeout%:61000:30000)"
> ;dialcommand_param = "|60|gL(%timeout%)"
> dialcommand_param = "|60|gS(%timeout%)"
> ;dialcommand_param = "|60|g"
>
> ; by default (3600000  =  1HOUR MAX CALL)
> dialcommand_param_sipiax_friend = "|60|HRgirL(3600000:61000:30000)"
>
> ; Define the order to make the outbound call
> ; YES ->  SIP/dialedphonenumber at gateway_ip - NO
> SIP/gateway_ip/dialedphonenumber
> ; Both should work exactly the same but i experimented one case when
> gateway was supporting dialedphonenumber at gateway_ip
> ; So in case of trouble, try it out
> switchdialcommand = yes
>
> ; failover recursive search - define how many time we want to authorize
> the research of the failover trunk when a call fails (value : 0 - 20)
> failover_recursive_limit = 2
>
> ; For free calls, limit the duration: amount in seconds
> maxtime_tocall_negatif_free_route = 5400
>
> ; Send a reminder email to the user when they are under min_credit_2call
> send_reminder = NO
>
> ; enable to monitor the call (to record all the conversations)
> ; value : YES - NO
> record_call = NO
>
> ; format of the recorded monitor file
> monitor_formatfile = gsm
>
> ; Force to play the balance to the caller in a predefined currency, to
> use the currency set for by the customer leave this field empty
> agi_force_currency =
>
> ; CURRENCY SECTION
> ; Define all the audio (without file extensions) that you want to play
> according to currency (use , to separate, ie
> "usd:prepaid-dollar,mxn:pesos,eur:Euro,all:credit")
> currency_association = usd:dollars,mxn:pesos,eur:euros,all:credit
>
> ; Please enter the file name you want to play when we prompt the calling
> party to enter the destination number
> ; file_conf_enter_destination = prepaid-enter-number-u-calling-1-or-011
> file_conf_enter_destination = prepaid-enter-dest
>
> ; Please enter the file name you want to play when we prompt the calling
> party to choose the prefered language
> ; file_conf_enter_menulang = prepaid-menulang
> file_conf_enter_menulang = prepaid-menulang2
>
> ; Define if you want to bill the 1st leg on callback even if the call is
> not connected to the destination
> callback_bill_1stleg_ifcall_notconnected = YES
>
>




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