[asterisk-users] a2billing for residential voip usage
Vardan Harutyunyan
hvardan71 at gmail.com
Tue Jun 15 02:35:26 CDT 2010
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
--
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
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 /<talk2ram at gmail.com>/* wrote:
>
>
> From: ram <talk2ram at gmail.com>
> Subject: Re: [asterisk-users] a2billing for residential voip usage
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> 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
> <landysaccount at yahoo.com </mc/compose?to=landysaccount at yahoo.com>>
> 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.
>
>
>
>
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