Tarificador+CDR

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antetodo un cordial saludo, mi problema es el siguiente;
como realizar un tarificador con el cdr que trae asterisk? puesto que alli se coloca la duracion de la llamada, asi que bastaria con agregar un valor a esa llamadas en el mismo cdr, es posible y si lo es como lo hago? supongo que se debe hacer manejando las bases de datos y con phpmyadmin

estoy trabajando con trixbox y eso trae a2billing pero no encuentro la manera de tarificar con ese a2billing y mucho menos donde visualizarlo, aqui tambien se puede realizar lo del cdr ya que este archivo se puede exportar desde la interfaz web

tambien si existe alguna otra forma como hacer esto, hagamanmelo saber, cabe destacar que la tarificacion que deseo realizar es para las llamadas internas es decir las llamadas entre los anexos

Mira en que contexto esta

Imagen de RazaMetaL

Mira en que contexto esta configurado /etc/asterisk/a2billing.conf y mete a tus extensiones en ese contexto, no he probado a2billing para que registre el consumo interno, pero al menos puedes tener una estadistica de la utilizacion de las extensiones internas.

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No encuentro un contexto como tal!

no veo un constexto como tal;

aqui te envio el archivo de configuracion, de existir el contexto, como meto mis extensiones alli? no entiendo a que te refieres

por otra parte si haz tenido la oportunidad de manejar el reporte que te ofrece trixbox en la freepbx en la parte de REPORTE alli te ofrece el estado del CDR, en ese CDR se obtiene todo el reporte de las llamadas ejecutas, y alli se registra el tiempo de esa llamada, ese reporte en la parte inferior permite exportarlo bien sea como cvs o Pdf, siendo esto asi bastaria con modificar las tablas que son exportadas y mostradas en la interfaz grafica y agregar una columna en la cual se multiplique los segundos que duro la llamada por el precio que se le coloque al segundo! siendo esto otra opcion para poder obtener una tarificacion, como lo puedo hacer puesto a que me parece que eso se hace editando los archivos.PHP.

saludos y gracias

; config file for the A2Billing Callingcard platform

; Global Database Setup

[database]
hostname=localhost
port=5432
user=a2billinguser
password=a2billing
dbname=mya2billing
dbtype=mysql

; configuration for the Web interface
[webui]

; Path to store the asterisk configuration files
buddyfilepath = /etc/asterisk/

; Email of the admin (not used yet)
email_admin = info@areski.net

; Card lenght
len_cardnumber = 10

; Voucher lenght
len_voucher = 15

;amount of MOH class you have created in musiconhold.conf : acc_1, acc_2... acc_10 class etc...
num_musiconhold_class = 10

;MANAGER CONNECTION PARAMETERS
manager_host = localhost
manager_username = a2billinguser
manager_secret = a2billing

; Allow to display the help section inside the admin interface (YES - NO)
show_help="YES"

; Parameter of the upload
; PLEASE CHECK ALSO THE VALUE IN YOUR PHP.INI THE LIMIT IF 2MG BY DEFAULT
my_max_file_size_import = 512000
my_max_file_size = 512000 ; in bytes

; Not used yet, goal is to upload files and use them directly in the IVR
dir_store_audio = /var/lib/asterisk/sounds/a2billing

;Parameter of the upload
my_max_file_size_audio=3072000 ; in bytes

; the file type extensions allowed to be uploaded such as "gsm, mp3, wav" (separate by ,)
file_ext_allow = gsm, mp3, wav

; the file type extensions allowed to be uploaded for the musiconhold such as "gsm, mp3, wav" (separate by ,)
file_ext_allow_musiconhold = mp3

; ENABLE THE CDR VIEWER TO LINK ON THE MONITOR FILES (YES - NO)
link_audio_file = "NO"

; PATH TO LINK ON THE RECORDED MONITOR FILES
monitor_path = /var/spool/asterisk/monitor
// grant access to apache user on read mode for the directory :> chmod 755 /var/spool/asterisk/monitor/

; FORMAT OF THE RECORDED MONITOR FILE
monitor_formatfile = gsm

; Display the icon in the invoice
show_icon_invoice = "YES"

; Display the top frame (useful if you want to save space on your little tiny screen )
show_top_frame = "YES"

;base currency define the default currency that you want to use to setup your system (see the file /etc/asterisk/rates.inc to know the currency code)
base_currency = usd

; currency_choose allow you to great a set of currencies to let the customer select the most appropriate ("all" can be used)
currency_choose = usd, eur, cad, hkd

; configuration for the Reccurring process (cront)
[recprocess]
batch_log_file=/tmp/batch-a2billing.log

; configuration for the AGI, different configuration can be defined, ie "agi-conf1", "agi-conf2", etc...
; the groupid parameter will define which process_sections to use. Usage : DeadAGI(a2billing.php|%groupid%)
; by default agi-conf1 is used
[agi-conf1]

; the debug level
; 0=none, 1=low, 2=normal, 3=all
debug=0

; Active the logging of the application
; logging is optimized to write all the logs at once :D
logger_enable=YES

; File to log
log_file=/tmp/a2billing.log

; if YES Use Set(LANGUAGE()=fr) instead, for me it didnt work from AGI
; ### if (SETLANGUAGE_DEPRECATE==YES) $myres = $agi->agi_exec("EXEC Set('LANGUAGE()=$language')");
setlanguage_deprecate=YES

; play the goodbye message when the user finish
say_goodbye=NO

; enable the menu to choose the language
; press 1 for English, pulsa 2 para el español, Pressez 3 pour Français
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 install)
force_language=

; Introduction prompt : to specify an additional prompt to play at the beginning of the application
; parlezplus-intro_013centimes
intro_prompt=

; lenght of the cardnumber (amount of digits)
len_cardnumber=10

; Voucher lenght
len_voucher = 15

; this is the minimum amount of credit to use the application
min_credit_2call=0

; if YES it will catch the DNID and try to dial it out directly without asking for the phonenumber to call
; value : YES, NO
use_dnid=NO

; list the dnid on which you want to avoid the use of the previous option "use_dnid"
no_auth_dnid=2400,2300

;number of time the user can dial different number
number_try=3

; Play the balance to the user after the authentication (values : yes - no)
say_balance_after_auth=YES

; Play the balance to the user after the call (values : yes - no)
say_balance_after_call=NO

; Play the time the user can call (values : yes - no)
say_timetocall=YES

; enable the callerid authentication
; if this option is active the CC system will check the CID of caller
cid_enable=NO

; if the cid doesnt exist you can then ask a cardnumber to the calling party in order to authenticate the caller
cid_askpincode_ifnot_callerid=YES

; if the callerID, this option will allow the system to add it automatically and create a cardnumber to hook them up.
cid_auto_create_card=NO

; If cid_auto_create_card has been set to YES, the following option will define with which parameters the card will be create
;
; 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 here 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

; enable the option to call sip/iax friend for free (values : YES - NO)
sip_iax_friends=NO

; if SIP_IAX_FRIENDS is active, you define a prefix for the dialed phonenumber to call directly a pstn number
; values : number
sip_iax_pstn_direct_call_prefix=9

; this will enable a prompt to enter your destination number_try
; 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

; 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
; 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="|30|HL(%timeout%:61000:30000)"

; by default (3600000 = 1HOUR MAX CALL)
dialcommand_param_sipiax_friend="|30|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 troubles, try it out
switchdialcommand=NO

; enable to monitor the call (to record all the conversation)
; value : YES - NO
record_call=NO

; format of the recorded monitor file
monitor_formatfile=gsm

;base currency define the default currency that you want to use to setup your system (see the file /etc/asterisk/rates.inc to know the currency code)
base_currency = usd

; 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 extension) that you want to play according to currency (use , to separate, ie "usd:prepaid-dollar,mxn:pesos,eur:Euro,all:credit")
currency_association = usd:prepaid-dollar,mxn:pesos,eur:euro,all:credit

; Please enter here the file you want to play when we prompt the calling party to enter his destination number
; file_conf_enter_destination = prepaid-enter-number-u-calling-1-or-011
file_conf_enter_destination = prepaid-enter-dest

; Please enter here the file 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

; the debug shell (ONLY FOR THE DEVELOPERS)
; 0=no, 1=yes
debugshell=0

Yo tengo esta opcion: ;

Imagen de RazaMetaL

Yo tengo esta opcion:


; asterisk context for sip and iax clients ( default: a2billing )
context=a2billing

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