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/dialedphonenum...@gateway_ip - NO SIP/gateway_ip/dialedphonenumber ; Both should work exactly the same but i experimented one case when gateway was supporting dialedphonenum...@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 -- 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: [email protected] www.eif-it.com Landy Landy wrote: > Ram. > Thanks for replying. I have searched / googled about it but can't find a > solution to monitor the 4 extensions I have at home. A2billing asks for > the number I want to dial but, I don't need that. I would like the > extensions to dial out normally and a2billing just record the time and > talked time for later review. > > Thanks. > > --- On *Tue, 6/15/10, ram /<[email protected]>/* wrote: > > > From: ram <[email protected]> > Subject: Re: [asterisk-users] a2billing for residential voip usage > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Date: Tuesday, June 15, 2010, 1:05 AM > > you see lot of documentation on wiki > Google them many success case you see > Ram > > On Tue, Jun 15, 2010 at 7:01 AM, Landy Landy > <[email protected] </mc/[email protected]>> > wrote: > > Hello List. > > I just installed a2billing with asterisk 1.6 and got it working. > The only problem is that I'm trying to setup something to manage > who's using the most minutes in the house. I noticed a2billing > only works for callin cards setups, or maybe I didn't configure > it correctly for what I want. Can I use a2billing for "•VoIP > residential services"? if yes, how? if no, please guide me to > another application I can use along side asterisk. > > Thanks in advanced for your time. > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by > http://www.api-digital.com <http://www.api-digital.com/> -- > New to Asterisk? Join us for a live introductory webinar every > Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -----Inline Attachment Follows----- > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
