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 Post subject: A2Billing Not Allowing Land Line Calls
PostPosted: Thu Mar 08, 2007 4:59 pm 
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Joined: Thu Mar 08, 2007 4:36 pm
Posts: 11
I've just installed asterisk 1.4 and a2billing 1.2.3 on RHEL4. I'm having this issue that when I call the asterisk box that the a2b prompt will only allow me to dial a SIP or IAX 'friend'. To be specific the gsm file that's played states, "Please enter the number of the SIP or IAX friend you wish to call, and then press the pound key.". Whenever I enter an actual phone number I get, "There is none SIP or IAX friend that match with this number".

How do I setup a2b to replace this prompt to allow me to enter an actual phone number (ie 1-800-555-5555), and connect to it? Any help/information would be greatly appreciated. Thanks in advance.


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 Post subject: A2Billing Not Allowing Land Line Calls
PostPosted: Fri Mar 09, 2007 12:39 pm 
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Joined: Thu Mar 08, 2007 4:36 pm
Posts: 11
Here's my a2billing.conf code to help out a little:
====================================
; config file for the A2Billing Callingcard platform



; Global Database Setup - select the database type and authentication as required.

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


; CALL-BACK
[callback]
; When web call-back is enabled this is the context to sent the call.
context_callback=a2billing-callback

; this is the Extension to redirect the call when the web callback is returned
extension=1000

; this is the number of seconds to wait before initiating the call back.
sec_wait_before_callback=10

;Number of seconds before the call-back can be re-initiated from the web page
; to prevent repeated and unwanted calls.
sec_avoid_repeate=30

; if the callback doesnt succeed within the value below, then the call is deemed to have failed.
timeout=20

; if we want to manage the answer on the call
answer_call=yes


; PREDICTIVE DIALER
; number of calls an agent will do when the call button is clicked
nb_predictive_call=10

; Number of days to wait before the number becomes available to call again.
nb_day_wait_before_retry=1

; The context to redirect the call for the predictive dialer
context_preditctivedialer=a2billing-predictivedialer


; When a call is made we need to limit the call duration : amount in seconds
predictivedialer_maxtime_tocall = 5400

; set the callerID for the predictive dialer and call-back
callerid = 123456

; ID Tariffgroup to use when you use the all-callback mode, check the ID in the tariffgroup list - WebUI
all_callback_tariff=1



; CUSTOMISATION Of THE CUSTOMER INTERFACE

[webcustomerui]
; Enable display of the payment interface - yes or no
customerinfo=yes

; Enable the Call history - yes or no
cdr=yes

; Enable invoices - yes or no
invoice=yes

; Enable the voucher screen - yes or no
voucher=yes

; Enable the paypal payment buttons - yes or no
paypal=yes

; Allow Speed Dial capabilities - yes or no
speeddial=yes

; Enable the DID (Direct Inwards Dialling) interface - yes or no
did=yes

; Show the ratecards - yes or no
ratecard=yes

; Offer simulator option on the customer interface - yes or no
simulator=yes

; Enable the callback option on the customer interface - yes or no
callback=yes

; Enable the predictivedialer option on the customer interface - yes or no
predictivedialer=yes

; Let users use SIP/IAX Webphone (Options : yes/no)
webphone=yes

;IP address or domain name of asterisk server that would be used by the web-phone
webphoneserver=localhost

; Let the users add new callerid
callerid=yes

; Let the user change the webui password
password=yes

; The total number of callerIDs for CLI Recognition that can be add by the customer
limit_callerid=5

; Email address to send the notification and error report - new DIDs assigned will also be emailed.
[email protected]



; SIGNUP MODULE
[signup]
; enable the signup module
enable_signup = 1

; amount of credit applied to a new user.
credit=0

; ID Tariffgroup to use, check the ID in the tariffgroup list - WebUI
tariff=1

; Specify whether the card is created as active or pending
activated=no

; Simultaneous or non concurrent access with the card - 0=INDIVIDUAL ACCESS or 1=SIMULTANEOUS ACCESS
simultaccess=0

;PREPAID CARD = 0 - POSTPAY CARD = 1
typepaid=0

; Define credit limit, which is only used for a POSTPAY card.
creditlimit=0

; Authorise the recurring service to apply on this card - Yes 1 - No 0
runservice=0

; Enable the expiry of the card - Yes 1 - No 0
enableexpire=0

; Expiry Date format YYYY-MM-DD HH:MM:SS. For instance, '2004-12-31 00:00:00'
expirationdate=

The number of days after which the card will expire
expiredays=0

; Create a sip account from signup ( default : yes )
sip_account=yes

; Create an iax account from signup ( default : yes )
iax_account=yes

; Set asterisk amaflag for sip and iax clients ( default: billing )
amaflag=billing

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

; active card after the new signup. if No, the Signup confirmation is needed and an email will be sent
; to the user with a link for activation (need to put the link into the Signup mail template)
activatedbyuser=no

; url of the customer interface to display after activation
urlcustomerinterface=http://asterisk-build.algomau.ca/a2customer/

; PAYPAL SETTINGS
[paypal]

Set the Paypal Email account.
; paypal_email=user@locahost

; Set the from:address for the transaction notification
[email protected]

; Define the from:name for the transaction notification
from_name=Paypal

; Set the company name to send for the transaction notification
company_name=A2Billing

; Set the email address for the notification and error report
[email protected]

; The Item name that is displayed to the customer when credit is purchased
item_name="Credit Purchase"

; Currency for the purchase, only one currency can be defined
currency_code=CAD

;IMPORTANT - Define here the URL to notify the payment back to the customer
;note that this is initiated by Paypal, and uses port 445 HTTPS, so this port must be open in the firewall.
notify_url=http://asterisk-build.algomau.ca/A2Billing/paypal/ipn_res.php

;Set the different amounts which are available - 5 amounts maximum (5:10:15)
purchase_amount=10:20:50

; paypal log file, to log the transactions & errors
paypal_logfile=/tmp/a2billing_paypal.log




;BACK-UP AND RESTORE
; configuration for backup and restore
[backup]

; Path to store backup of database
backup_path = /tmp

; path for gzip
gzip_exe=/bin/gzip

; path for gunzip
gunzip_exe=/bin/gunzip

; path for mysqldump
mysqldump=/usr/bin/mysqldump

; path for pg_dump
pg_dump=/usr/bin/pg_dump

; path for mysql
mysql=/usr/bin/mysql

;path for psql
psql=/usr/bin/psql



; WEB INTERFACE AND API CONFIGURATION
[webui]

; Path to store the asterisk configuration files SIP & IAX
buddy_sip_file = /etc/asterisk/additional_a2billing_sip.conf
buddy_iax_file = /etc/asterisk/additional_a2billing_iax.conf

; API have a security key to validate the http request, the key has to be sent after applying md5
; Valid characters are [a-z,A-Z,0-9]
api_security_key = Ae87v56zzl34v

; API to restrict the IP's authorised to make a request.
; Define The the list of ips separated by ;
api_ip_auth = 127.0.0.1

; Log file to store the ecommerce API requests
api_logfile = /tmp/api_ecommerce_request.log

; Log file to store the SOAP API requests
soap_logfile = /tmp/api_soap_request.log

; Administative Email(not used yet)
email_admin = [email protected]

; Card length
len_cardnumber = 10

; Alias-Card length
len_aliasnumber = 15

; Voucher length
len_voucher = 15

;MOH (Music on Hold) base directory
dir_store_mohmp3 = /var/lib/asterisk/mohmp3

;Number of MOH classes 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 = myasterisk
manager_secret = mycode

; Display the help section inside the admin interface (YES - NO)
show_help = YES

; File Upload parameters
; PLEASE CHECK ALSO THE VALUE IN YOUR PHP.INI THE LIMIT IS 2MG BY DEFAULT
my_max_file_size_import = 512000
my_max_file_size = 512000 ; in bytes

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

; upload maximum file size
my_max_file_size_audio=3072000 ; in bytes

; File type extensions permitted to be uploaded such as "gsm, mp3, wav" (separated by ,)
file_ext_allow = gsm, mp3, wav

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


; RECORDED CONVERSATIONS

; Enable link on the CDR viewer to the recordings. (YES - NO)
link_audio_file = NO


; Path to link 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

;CURRENCY AND GENERAL SETTINGS

; Display the top frame (useful if you want to save space on your little tiny screen )
show_top_frame = NO

;base currency define the default currency that you want to use to setup your system.
base_currency = usd

; Allow the customer to chose the most appropriate currency ("all" can be used)
currency_choose = usd, eur, cad, hkd

; field to export in csv format from cc_card table
card_export_field_list = creationdate, username, credit, lastname, firstname

; field to export in csv format from cc_voucher table
voucher_export_field_list = id, voucher, credit, tag, activated, usedcardnumber, usedate, currency

; Advanced mode - Display additional configuration options on the ratecard (progressive rates, musiconhold, ...)
advanced_mode = NO



; RECURRING PROCESS(cront)

[recprocess]
;Set the location of the log file.
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=1


; Manage the answer on the call
answer_call=yes

; Activate application logging
; logging is optimized to write all the logs at once :D
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=1


; 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=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 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=YES

; 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=YES


; 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=PREPAY
; amount of credit of the new card
cid_auto_create_card_credit=10

; 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=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=0

; 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


; 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 retu
rn 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=YES


; 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 = cad

; 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


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 Post subject:
PostPosted: Sat Mar 10, 2007 3:11 pm 
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Joined: Wed Mar 15, 2006 8:46 am
Posts: 109
Quote:
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



try set "sip_iax_pstn_direct_call = Yes"


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 Post subject: A2Billing Not Allowing Land Line Calls
PostPosted: Mon Mar 12, 2007 6:30 pm 
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Joined: Thu Mar 08, 2007 4:36 pm
Posts: 11
Tried it...no change. Thanks for getting back to me though. Anything else you can think of?

What else besides a2billing.conf do you have to customize to get things working?


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 Post subject:
PostPosted: Mon Mar 12, 2007 11:32 pm 
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Joined: Fri Dec 15, 2006 8:01 pm
Posts: 72
; 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=0

change to :

; 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

i compared your a2billing.conf to mine thats the only difference

Joe


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 Post subject: A2Billing Not Allowing Land Line Calls
PostPosted: Tue Mar 13, 2007 2:01 pm 
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Joined: Thu Mar 08, 2007 4:36 pm
Posts: 11
Didn't work. It's just so weird...everything I've read said that this shouldn't even be happening.

Most people have to struggle to get the IAX/SIP friend thing going, and I can't get mine to go away...GO FIGURE. :)

Anything else that anyone can think of.

Added after 2 hours 10 minutes:

I'm beyond frustrated...I'm starting from scratch. :x


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 Post subject:
PostPosted: Tue Mar 13, 2007 2:19 pm 
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Joined: Tue Jun 06, 2006 12:14 pm
Posts: 685
Location: florida
Just a guess if you have the other items set proper (sip_iax_pstn_direct_call = Yes) ... but perhaps do you not have your routes set properly?

I don't know what system would do if you do not have a route for the PSTN call, perhaps it knows there are zero routes for it and won't even prompt you that way. Try to do a call with the CLI and read it to see if you get a message like "rate engine finds no rate/route" or something like that.


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 Post subject: FIXED
PostPosted: Mon Mar 26, 2007 10:40 pm 
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Joined: Mon Mar 26, 2007 9:57 pm
Posts: 3
Location: bahia blanca - argentina
Hi,

I've the same problem.
When I make a call, I get this CLI logs:

Code:
asterisk1*CLI>
asterisk1*CLI>
  a2billing.php: line:705 - RES DTMF : XXYYZZTTUU
  a2billing.php: line:713 - SIP o IAX DESTINATION : XXYYZZTTUU
  a2billing.php: line:720 - SELECT name FROM cc_iax_buddies, cc_card WHERE cc_iax_buddies.name=cc_card.username AND useralias='XXYYZZTTUU'
  a2billing.php: line:722 - 0
  a2billing.php: line:729 - SELECT name FROM cc_sip_buddies, cc_card WHERE cc_sip_buddies.name=cc_card.username AND useralias='XXYYZZTTUU'
  a2billing.php: line:731 - 0
  a2billing.php: line:166 - [CHANNEL STATUS : 6 = Line is up]
  a2billing.php: line:170 - [CREDIT STATUS : 10.00000]
  a2billing.php: line:171 - [CREDIT MIN_CREDIT_2CALL : 0]
  a2billing.php: line:238 - RES sip_iax_pstndirect_call DTMF : -1
  a2billing.php: line:252 - TRUNK - dnid : -1 (1)
  a2billing.php: line:267 - CALL SIP_IAX_BUDDY
    -- Playing 'prepaid-sipiax-enternumber' (language 'es')
  a2billing.php: line:705 - RES DTMF : -1
  a2billing.php: line:519 - UPDATE cc_card SET inuse=inuse-1 WHERE username='2853800006'
  == Spawn extension (custom-callingcard, XXYYZZTTUU, 3) exited non-zero on 'SIP/2853800006-089932f8'
asterisk1*CLI>



Seems like a2billing don't try to call forward through Trunks.
It only look for destination call on my "friends" sip or iax.

I've trixbox 2.0 and a2b 1.2.3


Best regards.
Andres.

Added after 33 minutes:


I've fixed problem putting on my a2billing.conf, next:


Code:
sip_iax_friends=NO


regards,


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 Post subject: No call to Land Line
PostPosted: Wed Apr 25, 2007 7:11 am 
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Joined: Tue Apr 17, 2007 5:00 am
Posts: 2
Just went to a2billing user inter face.In define Rate Card Just enter 'defaultprefix' to DIALPREFIX field.


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 Post subject: Re: A2Billing Not Allowing Land Line Calls
PostPosted: Thu Sep 20, 2007 1:05 pm 
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Joined: Wed Sep 19, 2007 1:56 pm
Posts: 3
in extensions_trixbox.conf

; un-comment the 6 lines below to work on incoming DIDs
[custom-callingcard]
exten => s,1,Answer
exten => s,2,Wait,2
exten => s,3,DeadAGI,a2billing.php
exten => s,4,Wait,2
exten => s,5,Hangup

set up an Inbound Route to direct the calls to "custom-callingcard,s,1"


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