provided by Star2Billing S.L.

a2billing.php: line:544 - 1 && && 5&& 0
a2billing.php: line:571 - DESTINATION ::> 10000
a2billing.php: line:573 - APPLY_RULES DESTINATION ::> 10000
a2billing.php: line:610 - ERROR ::> RateEngine didnt succeed to match the dialed number over the ratecard (Please check : id the ratecard is well create ; if the removeInter_Prefix is set according to your prefix in the ratecard ; if you hooked the ratecard to the tariffgroup)

Can someone explain me why still enabling the option to call sip/iax friend for free my a2billing its searching a Rate to apply to my FREE call?

Thnx in advance

Did you dial through 9 alias number? Do you have free dialing and dialing to sip/iax friends in a2billing.conf file??

[agi-conf1]

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


; Manage the answer on the call
answer_call=yes

; Activate application logging
; logging is optimized to write all the logs at once
logger_enable=YES

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


; 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 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 installed)
force_language=

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

; length of the cardnumber (number of of digits)
len_cardnumber=10

; Alias-Card length
len_aliasnumber = 15

; Voucher length
len_voucher = 15

; 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=YES

; if notenoughcredit_cardnumber = YES then assign the CallerID to the new cardnumber
notenoughcredit_assign_newcardnumber_cid=YES


; 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=3


; 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 presentation of a callerID number
auto_setcallerid=YES

; If set, and auto_setcallerid is enabled, the number is sent as CID always
force_callerid=

; If force_callerid is not set, then this ensures that CID is set to one of the ccard'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=yes

; if the CID does not exist, you can then ask for a cardnumber from 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 the callerID authenticate is on, this option will allow the assign the cardnumber enter to the callerID if the callerID wasnt in the DB
cid_auto_assign_card_to_cid=yes

; 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 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

; if we want to check the 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=YES

; 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=9

; 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=YES


; 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=


; 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 trouble, try it out
switchdialcommand=no


; 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


;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 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:prepaid-dollar,mxn:pesos,eur:euro,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

i went to line 610 in a2billing.php trying to know whats going on with this error:
a2billing.php: line:610 - ERROR ::> RateEngine didnt succeed to match the dialed number over the ratecard (Please check : id the ratecard is well create ; if the removeInter_Prefix is set according to your prefix in the ratecard ; if you hooked the ratecard to the tariffgroup)

and what i found its this:

$failover_trunk = $RateEngine -> ratecard_obj[0][40+$usetrunk_failover];

may i setup an other trunk for sip to sip calls? What i understand... its that a2billing its trying to route my internal calls through my sip trunk... the one im using to make external calls. i think im lost

try this

1. look up the "alias" for the sip-friend you are going to call
2. dial 9+alias

where to find the alias? look in the customer record

gue is right, your system think this is a calling card. make sure your sip friend is register with your box - show sip peers.
Dialing 9 alias should get the job done.