[Asterisk-Users] A2billing problem.The system disconnects me
immediatelly after asking me the PIN
Administrator TOOTAI
admin at tootai.net
Fri Nov 11 03:22:46 MST 2005
Bukoka Budoka a écrit :
> Hi,
>
> i installed A2billing according to instructions. GUI works fine, i
> entered a new card and i put the appopriate context in my
> extensions.conf:
>
> [callingcard]
> exten => _123,1,Answer
> exten => _123,2,Wait,2
> exten => _123,3,DeadAGI,a2billing.php
> exten => _123,4,Wait,2
> exten => _123,5,Hangup
>
> However when i enter the Calling card system, it did not ask my to put
> my account-code. I found that in the a2billing.conf file there is a
> option for the intro_prompt which was empty. I put there the
> prepaid-enter-card-num gsm file and now when i enter the calling card
> system i have the following behavior:
>
> It asks me to enter the prepaid card number but immediatelly after it
> disconnects me with a message of "authentication failed - goodbye"...
>
> Why the system does not "wait" for the PIN number to be entered?
cid_enable=yes
Daniel
>
> Any ideas?
>
> Thank you,
>
> Budoka.
>
> ++++++++++++++++++++++++++++++++++++++++++++++++++++++++
>
> My a2billing.conf is as follows:
>
> ; config file for the A2Billing Callingcard platform
> ; Global Database Setup
>
> [database]
> hostname=localhost
> port=5432
> user=a2billinguser
> password=a2billing
> dbname=mya2billing
> ;dbtype=postgres
> 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 at areski.net
>
> ; Card lenght
> len_cardnumber = 4
>
> ; Voucher lenght
> len_voucher = 5
>
> ;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 = panos
> manager_secret = panos123
>
> ; 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=3
>
>
> ; 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=YES
>
> ; 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=EN
>
> ; Introduction prompt : to specify an additional prompt to play at the
> beginning of the application
> ; parlezplus-intro_013centimes
> intro_prompt=prepaid-enter-card-num
>
> ; lenght of the cardnumber (amount of digits)
> len_cardnumber=4
> ; Voucher lenght
> len_voucher = 5
>
> ; this is the minimum amount of credit to use the application
> min_credit_2call=1
>
> ; 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 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 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
>
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