<p>Pruonckk, você tem certeza que é o caso? Quando comento as linhas do asterisk dentro do from-internal ele passa a transferir normalmente.(preferi fazer assim do que criar o extensions_a2billing.conf)</p>
<p>Ou seja, tirou o contexto a2billing, tudo funciona normalmente.</p>
<p>Você teria como esclarecer melhor?</p>
<p>Obrigado pela ajuda!</p>
<p>João Queiroz</p>
<p> </p>
<p> </p>
<p>Em 27/04/2009 18:22, <strong><span>pruonckk@pruonckk.org</span></strong> escreveu:</p>
<blockquote style="border-left: 2px solid #6868cc; margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br /><br />o transfer voce tem que por o parametro t ou T (ou os dois) , abaixo<br />descrição<br /><br /># t: Allow the called user to transfer the call by hitting the blind xfer<br />keys (features.conf)<br /><br /> * If you have set the variable GOTO_ON_TRANSFER then the transferrer<br />will be sent to the context|exten|pri (you can use ^ to represent | to<br />avoid escapes), example: SetVar(GOTO_ON_TRANSFER=woohoo^s^1); works<br />with both t and T<br /> * WARNING: GOTO_ON_TRANSFER does not exist in any version of ASTERISK<br />and will not! the variable is called GOTO_ON_BLINDXFR see<br />http://svn.digium.com/view/asterisk?rev=5495&view=rev and<br />http://bugs.digium.com/view.php?id=4056 for details. THX to the person<br />who shared the information above!<br /><br /># T: Allow the calling user to transfer the call by hitting the blind xfer<br />keys (features.co
nf)<br /><br /><br /><br />>
<p>Antes de postar aqui eu dei uma boa googleada e encontrei algo sobre o<br />> parâmetro "i" no dialcommand porém não obtive sucesso, abaixo segue<br />> meus parâmetros de discagem. Se tive paciência segue também meu<br />> a2billing.conf completo.</p>
<br />>
<p>dialcommand_param = "|60|HRgrL(%timeout%:61000:30000)"</p>
<br />>
<p>; by default (3600000 = 1HOUR MAX CALL)<br /><br />> />dialcommand_param_sipiax_friend = "|60|HRgirL(3600000:61000:30000)"</p>
<br />>
<p>; Define the order to make the outbound call<br />; YES -><br />> SIP/dialedphonenumber@gateway_ip - NO SIP/gateway_ip/dialedphonenumber<br /><br />> />; Both should work exactly the same but i experimented one case when<br />> gateway was supporting dialedphonenumber@gateway_ip <br />; So in case of<br />> trouble, try it out<br />switchdialcommand = NO</p>
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<p>Â</p>
<br />>
<p>Tentei também alterar o contexto do a2billing para o descrito aqui:<br />> http://forum.asterisk2billing.org/viewtopic.php?f=16&t=3044&start=15<br />> também sem sucesso.</p>
<br />>
<p>Â</p>
<br />>
<p>Minha instalação roda no trixbox e os ramais foram importados usando<br />> o bulk_extensions.</p>
<br />>
<p>Â</p>
<br />>
<p>Mais alguma idéia?</p>
<br />>
<p>Segue meu a2billing.conf.</p>
<br />>
<p>Obrigado,</p>
<br />>
<p>João Queiroz</p>
<br />>
<p>______________________________</p>
<br />>
<p>Â</p>
<br />>
<p>;<br />; config file for the A2Billing Callingcard platform<br />;</p>
<br />>
<p><br />; Global Database Setup - select the database type and<br />> authentication as required.</p>
<br />>
<p>[database]<br />hostname = localhost<br />port = 5432<br />user =<br />> a2billinguser<br />password = a2billing<br />dbname = mya2billing<br /><br />> />;dbtype = postgres<br />dbtype = mysql</p>
<br />>
<p><br />[global]<br />; len_cardnumber is removed<br />; interval for the<br />> length of the cardnumber (number of digits), minimum lenght is 4<br />;<br />> ie: 10-15 (cardnumber authorised 10, 11, 12, 13, 14, 15) ; 10,12,14<br />> (cardnumber authorised 10, 12, 14)<br />interval_len_cardnumber = 10</p>
<br />>
<p>; Alias-Card length<br />len_aliasnumber = 10</p>
<br />>
<p>; Voucher length<br />len_voucher = 10</p>
<br />>
<p>;base currency define the default currency that you want to use to<br />> setup your system (see the currency table to know the currency code)<br /><br />> />base_currency = brl</p>
<br />>
<p>; filename of the image that will be display at the top of the invoice<br />> (if not defined no image will appear ; path to place the image<br />> templates/default/images/)<br />; the type of file have to be a<br />> jpeg/jpg<br />invoice_image = asterisk01.jpg</p>
<br />>
<p>; DID Billing - amount of day before the end of the monthly reservation<br />> to bill the customer to for the DID use<br />; if the user dont have<br />> enough credit he will get an email asking him to refill<br /><br />> />didbilling_daytopay = 5</p>
<br />>
<p>;webiste administrator email address<br />admin_email =<br />> areski@gmail.com</p>
<br />>
<p>; MANAGER CONNECTION PARAMETERS<br />manager_host = localhost<br /><br />> />manager_username = a2billinguser<br />manager_secret = a2billing</p>
<br />>
<p><br />; CALL-BACK<br />[callback]<br />; When web call-back is enabled<br />> this is the context to sent the call.<br />context_callback =<br />> a2billing-callback</p>
<br />>
<p>; this is the Extension to redirect the call when the web callback is<br />> returned<br />extension = 1000</p>
<br />>
<p>; this is the number of seconds to wait before initiating the call<br />> back.<br />sec_wait_before_callback = 10</p>
<br />>
<p>;Number of seconds before the call-back can be re-initiated from the<br />> web page<br />; to prevent repeated and unwanted calls. <br /><br />> />sec_avoid_repeate = 30</p>
<br />>
<p>; if the callback doesnt succeed within the value below, then the call<br />> is deemed to have failed.<br />timeout = 20</p>
<br />>
<p>; if we want to manage the answer on the call<br />; Disabling this for<br />> callback trigger numbers makes it ring not hang up.<br />answer_call =<br />> yes</p>
<br />>
<p><br />; PREDICTIVE DIALER<br />; number of calls an agent will do when<br />> the call button is clicked<br />nb_predictive_call = 10</p>
<br />>
<p>; Number of days to wait before the number becomes available to call<br />> again.<br />nb_day_wait_before_retry = 1</p>
<br />>
<p>; The context to redirect the call for the predictive dialer<br /><br />> />context_preditctivedialer = a2billing-predictivedialer</p>
<br />>
<p><br />; When a call is made we need to limit the call duration : amount<br />> in seconds <br />predictivedialer_maxtime_tocall = 5400</p>
<br />>
<p>; set the callerID for the predictive dialer and call-back<br /><br />> />callerid = 123456</p>
<br />>
<p>; ID Call Plan to use when you use the all-callback mode, check the ID<br />> in the "list Call Plan" - WebUI<br />all_callback_tariff = 1</p>
<br />>
<p>; Define the group of servers that are going to be used by the<br />> callback<br />id_server_group = 1</p>
<br />>
<p>; Audio intro message when the callback is initiate <br /><br />> />callback_audio_intro = prepaid-callback_intro</p>
<br />>
<p><br />; CUSTOMISATION Of THE CUSTOMER INTERFACE<br /><br />> />[webcustomerui]</p>
<br />>
<p>; url of the signup page to show up on the sign in page (if empty no<br />> link will show up)<br />signup_page_url =</p>
<br />>
<p>;Enable or disable the payment methods; yes for multi-payment or no for<br />> single payment method option<br />paymentmethod = no</p>
<br />>
<p>;Enable or disable the page which allow customer to modify its personal<br />> information<br />personalinfo = no</p>
<br />>
<p>; Enable display of the payment interface - yes or no<br />customerinfo<br />> = no</p>
<br />>
<p>; Enable display of the sip/iax info - yes or no<br />sipiaxinfo =<br />> no</p>
<br />>
<p>; Enable the Call history - yes or no<br />cdr = yes</p>
<br />>
<p>; Enable invoices - yes or no<br />invoice =no</p>
<br />>
<p>; Enable the voucher screen - yes or no<br />voucher = no</p>
<br />>
<p>; Enable the paypal payment buttons - yes or no<br />paypal = no</p>
<br />>
<p>; Allow Speed Dial capabilities - yes or no<br />speeddial = no</p>
<br />>
<p>; Enable the DID (Direct Inwards Dialling) interface - yes or no<br /><br />> />did = no</p>
<br />>
<p>; Show the ratecards - yes or no<br />ratecard = no</p>
<br />>
<p>; Offer simulator option on the customer interface - yes or no<br /><br />> />simulator = yes</p>
<br />>
<p>; Enable the callback option on the customer interface - yes or no<br /><br />> />callback = no</p>
<br />>
<p>; Enable the predictivedialer option on the customer interface - yes or<br />> no<br />predictivedialer = no</p>
<br />>
<p>; Let users use SIP/IAX Webphone (Options : yes/no)<br />webphone =<br />> yes</p>
<br />>
<p>;IP address or domain name of asterisk server that would be used by the<br />> web-phone<br />webphoneserver = localhost</p>
<br />>
<p>; Let the users add new callerid<br />callerid = no</p>
<br />>
<p>; Let the user change the webui password<br />password = yes</p>
<br />>
<p>; The total number of callerIDs for CLI Recognition that can be add by<br />> the customer<br />limit_callerid = 5</p>
<br />>
<p>; Email address to send the notification and error report - new DIDs<br />> assigned will also be emailed.<br />error_email =<br />> confidencial@confidencial.com</p>
<br />>
<p>; URL for specific return if an error occur after login<br /><br />> />return_url_distant_login =</p>
<br />>
<p>; URL for specific return if an error occur after forgetpassword<br /><br />> />return_url_distant_forgetpassword =</p>
<br />>
<p><br />;SIP & IAX client configuration information.<br /><br />> />[sip-iax-info]</p>
<br />>
<p>;Trunk Name to show in sip/iax info <br />sip_iax_info_trunkname =<br />> call-labs</p>
<br />>
<p>;Allowed Codec, ulaw, gsm, g729<br />; use multi value without spaces :<br />> "gsm,ulaw,g729"<br />sip_iax_info_allowcodec = g729</p>
<br />>
<p>;host information<br />sip_iax_info_host = call-labs.com</p>
<br />>
<p>;IAX Additional Parameters<br />iax_additional_parameters =<br />> "canreinvite = no"</p>
<br />>
<p>;SIP Additional Parameters<br />sip_additional_parameters = "trustrpid<br />> = yes | sendrpid = yes | canreinvite = no"</p>
<br />>
<p>[epayment_method]<br />enable = no<br />; eg, http://localhost -<br />> should not be empty for productive servers<br />http_server =<br />> "http://www.call-labs.com"<br />; eg, https://localhost - Enter here your<br />> Secure Server Address, should not be empty for productive servers<br /><br />> />https_server = "http://www.call-labs.com"<br />; Enter your Domain Name<br />> or IP Address, eg, 26.63.165.200<br />http_cookie_domain =<br />> 26.63.165.200<br />; Enter your Secure server Domain Name or IP Address,<br />> eg, 26.63.165.200<br />https_cookie_domain = 26.63.165.200<br />; Enter<br />> the Physical path of your Application on your server<br />http_cookie_path<br />> = "/A2BCustomer_UI/"<br />; Enter the Physical path of your Application on<br />> your Secure Server<br />https_cookie_path = "/A2BCustomer_UI/"<br />;<br />> Enter the Physical path of your Application on your server<br /><br />> />dir_ws_http_catalog =
"/A2BCustomer_UI/"<br />; Enter the Physical path<br />> of your Application on your Secure Server<br />dir_ws_https_catalog =<br />> "/A2BCustome<br />> r_UI/"<br />; secure webserver for checkout procedure?<br />enable_ssl =<br />> yes</p>
<br />>
<p>http_domain = 26.63.165.200</p>
<br />>
<p>dir_ws_http = "/~areski/svn/a2billing/payment/A2BCustomer_UI/"</p>
<br />>
<p>; maybe try with :<br />; Define here the URL to notify the payment<br /><br />> />; payment_notify_url=...</p>
<br />>
<p>;define the different amount of purchase that would be available - 5<br />> amount maximum (5:10:15)<br />purchase_amount = 1:2:5:10:20</p>
<br />>
<p>; Item name that would be display to the user when he will buy<br />> credit<br />item_name = "Credit Purchase"</p>
<br />>
<p>; Currency for the Credit purchase, only one can be define here<br /><br />> />currency_code = USD</p>
<br />>
<p>; Define here the URL of paypal gateway the payment (to test with<br />> paypal sandbox)<br />paypal_payment_url =<br />> "https://secure.paypal.com/cgi-bin/webscr"<br />;paypal_payment_url =<br />> "https://www.sandbox.paypal.com/cgi-bin/webscr"</p>
<br />>
<p>; paypal transaction verification url<br />paypal_verify_url =<br />> "ssl://www.paypal.com"<br />;paypal_verify_url =<br />> www.sandbox.paypal.com</p>
<br />>
<p>; Define here the URL of Authorize gateway <br />authorize_payment_url<br />> = "https://secure.authorize.net/gateway/transact.dll"<br /><br />> />;authorize_payment_url =<br />> "https://test.authorize.net/gateway/transact.dll"</p>
<br />>
<p>;paypal store name to show in the paypal site when customer will go to<br />> pay<br />store_name = Asterisk2Billing</p>
<br />>
<p>;Transaction Key for security of Epayment Max length of 60<br />> Characters.<br />transaction_key = asdf1212fasd121554sd4f5s45sdf</p>
<br />>
<p>;Moneybookers secret word<br />moneybookers_secretword = areski<br /><br />> <br />; SIGNUP MODULE<br />[signup]<br />; enable the signup module<br /><br />> />enable_signup = 1</p>
<br />>
<p>; enable Captcha on the signup module (value : YES or NO)<br /><br />> />enable_captcha = YES</p>
<br />>
<p>; amount of credit applied to a new user.<br />credit = 0</p>
<br />>
<p>; the list of id of call plans which will be shown in signup.<br /><br />> />callplan_id_list = 1, 2</p>
<br />>
<p>; Specify whether the card is created as active or pending<br /><br />> />activated = no</p>
<br />>
<p>; Simultaneous or non concurrent access with the card - 0 = INDIVIDUAL<br />> ACCESS or 1 = SIMULTANEOUS ACCESS<br />simultaccess = 0</p>
<br />>
<p>;PREPAID CARD = 0 - POSTPAY CARD = 1<br />typepaid = 0</p>
<br />>
<p>; Define credit limit, which is only used for a POSTPAY card. <br /><br />> />creditlimit = 999999999</p>
<br />>
<p>; Authorise the recurring service to apply on this card - Yes 1 - No<br />> 0<br />runservice = 0</p>
<br />>
<p>; Enable the expiry of the card - Yes 1 - No 0<br />enableexpire =<br />> 0</p>
<br />>
<p>; Expiry Date format YYYY-MM-DD HH:MM:SS. For instance, '2004-12-31<br />> 00:00:00' <br />expirationdate =</p>
<br />>
<p>; The number of days after which the card will expire <br />expiredays<br />> = 0</p>
<br />>
<p>; Create a sip account from signup ( default : yes )<br />sip_account =<br />> yes</p>
<br />>
<p>; Create an iax account from signup ( default : yes )<br />iax_account<br />> = yes</p>
<br />>
<p>; active card after the new signup. if No, the Signup confirmation is<br />> needed and an email will be sent <br />; to the user with a link for<br />> activation (need to put the link into the Signup mail template)<br /><br />> />activatedbyuser = no</p>
<br />>
<p>; url of the customer interface to display after activation<br /><br />> />urlcustomerinterface = http://localhost/A2BCustomer_UI/</p>
<br />>
<p>; Define if you want to reload Asterisk when a SIP / IAX Friend is<br />> created at signup time<br />reload_asterisk_if_sipiax_created = no</p>
<br />>
<p><br />;BACK-UP AND RESTORE<br />; configuration for backup and<br />> restore<br />[backup]</p>
<br />>
<p>; Path to store backup of database<br />backup_path = /tmp</p>
<br />>
<p>; path for gzip<br />gzip_exe = /bin/gzip</p>
<br />>
<p>; path for gunzip<br />gunzip_exe = /bin/gunzip</p>
<br />>
<p>; path for mysqldump<br />mysqldump = /usr/bin/mysqldump</p>
<br />>
<p>; path for pg_dump<br />pg_dump = /usr/bin/pg_dump</p>
<br />>
<p>; path for mysql<br />mysql = /usr/bin/mysql</p>
<br />>
<p>;path for psql<br />psql = /usr/bin/psql</p>
<br />>
<p>Â</p>
<br />>
<p>; WEB INTERFACE AND API CONFIGURATION<br />[webui]</p>
<br />>
<p>; Path to store the asterisk configuration files SIP & IAX<br /><br />> />buddy_sip_file = /etc/asterisk/additional_a2billing_sip.conf<br /><br />> />buddy_iax_file = /etc/asterisk/additional_a2billing_iax.conf</p>
<br />>
<p>; API have a security key to validate the http request, the key has to<br />> be sent after applying md5 <br />; Valid characters are [a-z,A-Z,0-9]<br /><br />> />api_security_key = Ae87v56zzl34v</p>
<br />>
<p>; API to restrict the IP's authorised to make a request. <br />; Define<br />> The the list of ips separated by ;<br />api_ip_auth = 127.0.0.1</p>
<br />>
<p>; Administative Email(not used yet)<br />email_admin =<br />> confidencial@confidencial.com</p>
<br />>
<p>; MOH (Music on Hold) base directory<br />dir_store_mohmp3 =<br />> /var/lib/asterisk/mohmp3</p>
<br />>
<p>; Number of MOH classes you have created in musiconhold.conf : acc_1,<br />> acc_2... acc_10 class etc...<br />num_musiconhold_class = 10</p>
<br />>
<p>; Display the help section inside the admin interface (YES - NO)<br /><br />> />show_help = YES</p>
<br />>
<p>; File Upload parameters<br />; PLEASE CHECK ALSO THE VALUE IN YOUR<br />> PHP.INI THE LIMIT IS 2MG BY DEFAULT<br />my_max_file_size_import = 1024000<br />> ; 1 MG</p>
<br />>
<p>; Not used yet, The goal is to upload files and use them in the IVR<br /><br />> />dir_store_audio = /var/lib/asterisk/sounds/a2billing</p>
<br />>
<p>; upload maximum file size<br />my_max_file_size_audio=3072000 ; in<br />> bytes</p>
<br />>
<p>; File type extensions permitted to be uploaded such as "gsm, mp3, wav"<br />> (separated by ,)<br />file_ext_allow = gsm, mp3, wav</p>
<br />>
<p>; File type extensions permitted to be uploaded for the musiconhold<br />> such as "gsm, mp3, wav" (separate by ,)<br />file_ext_allow_musiconhold =<br />> mp3</p>
<br />>
<p><br />; RECORDED CONVERSATIONS</p>
<br />>
<p>; Enable link on the CDR viewer to the recordings. (YES - NO)<br /><br />> />link_audio_file = yes</p>
<br />>
<p><br />; Path to link the recorded monitor files<br />monitor_path =<br />> /var/spool/asterisk/monitor<br />; grant access to apache user on read<br />> mode for the directory :> chmod 755 /var/spool/asterisk/monitor/</p>
<br />>
<p>; FORMAT OF THE RECORDED MONITOR FILE <br />monitor_formatfile =<br />> gsm</p>
<br />>
<p>; Display the icon in the invoice<br />show_icon_invoice = YES</p>
<br />>
<p>;CURRENCY AND GENERAL SETTINGS</p>
<br />>
<p>; Display the top frame (useful if you want to save space on your<br />> little tiny screen )<br />show_top_frame = NO</p>
<br />>
<p>; Allow the customer to chose the most appropriate currency ("all" can<br />> be used)<br />currency_choose = usd, eur, cad, hkd</p>
<br />>
<p>; field to export in csv format from cc_card table<br /><br />> />card_export_field_list = id, username, useralias, lastname, credit,<br />> tariff, activated, language, inuse, currency, sip_buddy, iax_buddy,<br />> nbused, mac_addr</p>
<br />>
<p>; field to export in csv format from cc_voucher table<br /><br />> />voucher_export_field_list = id, voucher, credit, tag, activated,<br />> usedcardnumber, usedate, currency</p>
<br />>
<p>; Advanced mode - Display additional configuration options on the<br />> ratecard (progressive rates, musiconhold, ...)<br />advanced_mode = NO</p>
<br />>
<p>; Delete the SIP/IAX Friend & callerid when a card is deleted<br /><br />> />delete_fk_card = yes</p>
<br />>
<p><br />; This section is basically used when we create a new friend <br /><br />> />; when you create a SIP IAX friend for a card the following parameters<br />> will define the default value for the peer creation<br />[peer_friend]<br /><br />> />; Refer to sip.conf & iax.conf documentation for the meaning of<br />> those parameters<br />; sip.conf -><br />> http://www.voip-info.org/wiki-Asterisk+config+sip.conf<br />; iax.conf<br />> -> http://www.voip-info.org/wiki-Asterisk+config+iax.conf<br />type =<br />> friend<br />allow = ulaw,alaw,gsm,g729<br />context = a2billing<br />; use<br />> "no" or "yes" with quote otherwise the value will be converted to 1 or<br />> 0<br />nat = "yes"<br />amaflag = billing<br />; use "no" or "yes" with<br />> quote otherwise the value will be converted to 1 or 0<br />qualify =<br />> "yes"<br />host = dynamic<br />dtmfmode = RFC2833</p>
<br />>
<p><br />[log-files]<br />; To disable application logging, remove/comment<br />> the log file name aside service</p>
<br />>
<p>; cront - recurring process <br />cront_alarm =<br />> /tmp/cront_a2b_alarm.log<br />cront_autorefill =<br />> /tmp/cront_a2b_autorefill.log<br />cront_batch_process =<br />> /tmp/cront_a2b_batch_process.log<br />cront_bill_diduse =<br />> /tmp/cront_a2b_bill_diduse.log<br />cront_subscriptionfee =<br />> /tmp/cront_a2b_subscription_fee.log<br />cront_currency_update =<br />> /tmp/cront_a2b_currency_update.log<br />cront_invoice =<br />> /tmp/cront_a2b_invoice.log<br />cront_check_account =<br />> /tmp/cront_a2b_check_account.log</p>
<br />>
<p>; paypal log file, to log all the transaction & error<br />paypal =<br />> /tmp/a2billing_paypal.log</p>
<br />>
<p>; epayment log file, to log all the transaction & error<br /><br />> />epayment = /tmp/a2billing_epayment.log</p>
<br />>
<p>; Log file to store the ecommerce API requests<br />api_ecommerce =<br />> /tmp/api_ecommerce_request.log</p>
<br />>
<p>; Log file to store the CallBack API requests<br />api_callback =<br />> /tmp/api_w<br />callback_request.log</p>
<br />>
<p>; File to log<br />agi = /tmp/a2billing_agi.log</p>
<br />>
<p>Â</p>
<br />>
<p>; configuration for the AGI, different configuration can be defined, ie<br />> "agi-conf1", "agi-conf2", etc...<br />; the groupid parameter will define<br />> which process_sections to use. Usage : DeadAGI(a2billing.php|%groupid%)<br /><br />> />; by default agi-conf1 is used<br />[agi-conf1]</p>
<br />>
<p>; the debug level<br />; 0=none, 1=low, 2=normal, 3=all<br />debug =<br />> 1</p>
<br />>
<p>; Asterisk Version Information<br />; 1_1,1_2,1_4 By Default it will<br />> take 1_2 or higher<br />asterisk_version = 1_2</p>
<br />>
<p>; Manage the answer on the call<br />answer_call = YES</p>
<br />>
<p>; Play audio - this will disable all stream file but not the Get Data<br />> <br />; for wholesale ensure that the authentication works and than<br />> number_try = 1<br />play_audio = YES</p>
<br />>
<p>; play the goodbye message when the user has finished.<br />say_goodbye<br />> = NO</p>
<br />>
<p>; enable the menu to choose the language<br />; press 1 for English,<br />> pulsa 2 para el español, Pressez 3 pour Français<br />play_menulanguage<br />> = NO</p>
<br />>
<p><br />; force the use of a language, if you dont want to use it leave<br />> the option empty<br />; Values : ES, EN, FR, etc... (according to the<br />> audio you have installed)<br />force_language = BR</p>
<br />>
<p>; Introduction prompt : to specify an additional prompt to play at the<br />> beginning of the application<br />intro_prompt =</p>
<br />>
<p>; Minimum amount of credit to use the application<br />min_credit_2call<br />> = 0</p>
<br />>
<p>; this is the minimum duration in seconds of a call in order to be<br />> billed<br />; any call with a length less than min_duration_2bill will<br />> have a 0 cost<br />; useful not to charge callers for system errors when a<br />> call was answered but it actually didn't connect<br />min_duration_2bill =<br />> 0</p>
<br />>
<p>; if user doesn't have enough credit to call a destination, prompt him<br />> to enter another cardnumber<br />notenoughcredit_cardnumber = YES</p>
<br />>
<p>; if notenoughcredit_cardnumber = YES then assign the CallerID to the<br />> new cardnumber<br />notenoughcredit_assign_newcardnumber_cid = NO</p>
<br />>
<p><br />; if YES it will use the DNID and try to dial out, without asking<br />> for the phonenumber to call<br />; value : YES, NO<br />use_dnid = YES</p>
<br />>
<p>; list the dnid on which you want to avoid the use of the previous<br />> option "use_dnid"<br />no_auth_dnid = 2400,2300</p>
<br />>
<p>; number of times the user can dial different number<br />number_try =<br />> 3</p>
<br />>
<p>; this will force to select a specific call plan by the Rate Engine<br /><br />> />force_callplan_id =</p>
<br />>
<p>; Play the balance to the user after the authentication (values : yes -<br />> no)<br />say_balance_after_auth = NO</p>
<br />>
<p>; Play the balance to the user after the call (values : yes - no)<br /><br />> />say_balance_after_call = NO</p>
<br />>
<p>; Play the initial cost of the route (values : yes - no)<br /><br />> />say_rateinitial = NO</p>
<br />>
<p>; Play the amount of time that the user can call (values : yes - no)<br /><br />> />say_timetocall = NO</p>
<br />>
<p><br />; enable the setup of the callerID number before the outbound is<br />> made, by default the user callerID value will be use<br />auto_setcallerid<br />> = YES</p>
<br />>
<p>; If auto_setcallerid is enabled, the value of force_callerid will be<br />> set as CallerID<br />force_callerid =</p>
<br />>
<p>; If force_callerid is not set, then the following option ensures that<br />> CID is set to one of the card's configured caller IDs or blank if none<br />> available.<br />; NO - disable this feature, caller ID can be anything.<br /><br />> />; CID - Caller ID must be one of the customers caller IDs<br />; DID -<br />> Caller ID must be one of the customers DID nos.<br />; BOTH - Caller ID<br />> must be one of the above two items.<br />cid_sanitize = NO</p>
<br />>
<p><br />; enable the callerid authentication<br />; if this option is<br />> active the CC system will check the CID of caller <br />cid_enable =<br />> NO</p>
<br />>
<p>; if the CID does not exist, then the caller will be prompt to enter<br />> his cardnumber<br />cid_askpincode_ifnot_callerid = YES</p>
<br />>
<p>; if the callerID authentication is enable and the authentication fails<br />> then the user will be prompt to enter his cardnumber<br />; this option<br />> will bound the cardnumber entered to the current callerID so that next<br />> call will be directly authenticate<br />cid_auto_assign_card_to_cid =<br />> NO</p>
<br />>
<p>; if the callerID is captured on a2billing, this option will create<br />> automatically a new card and add the callerID to it <br /><br />> />cid_auto_create_card = NO</p>
<br />>
<p>; set the length of the card that will be auto create (ie, 10)<br /><br />> />cid_auto_create_card_len = 10</p>
<br />>
<p>; If cid_auto_create_card has been set to YES, the following options<br />> will define with which configuration we will create the card<br />;<br />;<br />> billing type of the new card<br />; ( value : POSTPAY or PREPAY)<br /><br />> />cid_auto_create_card_typepaid = POSTPAY</p>
<br />>
<p>; amount of credit of the new card<br />cid_auto_create_card_credit =<br />> 0</p>
<br />>
<p>; if postpay, define the credit limit for the card<br /><br />> />cid_auto_create_card_credit_limit = 1000</p>
<br />>
<p>; the tariffgroup to use for the new card (this is the ID that you can<br />> find on the admin web interface)<br />cid_auto_create_card_tariffgroup =<br />> 6</p>
<br />>
<p>; to check callerID over the cardnumber authentication (to guard<br />> against spoofing)<br />callerid_authentication_over_cardnumber = NO</p>
<br />>
<p>; enable the option to call sip/iax friend for free (values : YES -<br />> NO)<br />sip_iax_friends = NO</p>
<br />>
<p>; if SIP_IAX_FRIENDS is active, you can define a prefix for the dialed<br />> digits to call a pstn number<br />; values : number<br /><br />> />sip_iax_pstn_direct_call_prefix = 555</p>
<br />>
<p>; this will enable a prompt to enter your destination number.<br />; if<br />> number start by sip_iax_pstn_direct_call_prefix we do directly a sip iax<br />> call, if not we do a normal call<br />sip_iax_pstn_direct_call = NO</p>
<br />>
<p>; enable the option to refill card with voucher in IVR (values : YES -<br />> NO)<br />ivr_voucher = NO</p>
<br />>
<p>; if ivr_voucher is active, you can define a prefix for the voucher<br />> number to refill your card<br />; values : number - don't forget to change<br />> prepaid-refill_card_with_voucher audio accordingly<br />ivr_voucher_prefix<br />> = 8</p>
<br />>
<p>; When the user credit are below the minimum credit to call<br />> min_credit<br />; jump directly to the voucher IVR menu (values: YES -<br />> NO)<br />jump_voucher_if_min_credit = NO</p>
<br />>
<p>; Extracharge DIDs, multiple numbers and fees must be separated by<br />> comma<br />; extracharge_did = 1800XXXXXXX,1888XXXXXXX<br /><br />> />extracharge_did = <br />;extracharge_fee = 0.02,0.03<br /><br />> />extracharge_fee = <br />;extracharge_buyfee = 0.015,0.025<br /><br />> />extracharge_buyfee =</p>
<br />>
<p>; List the prefixes that will be stripped off if the call plan requires<br />> it<br />international_prefixes = 011,00,09</p>
<br />>
<p>; More information about the Dial :<br />> http://voip-info.org/wiki-Asterisk+cmd+dial<br />; 30 : The timeout<br />> parameter is optional. If not specifed, the Dial command will wait<br />> indefinitely, exiting only when the originating channel hangs up, or all<br />> the dialed channels return a busy or error condition. Otherwise it<br />> specifies a maximum time, in seconds, that the Dial command is to wait for<br />> a channel to answer.<br />; H: Allow the caller to hang up by dialing *<br />> <br />; r: Generate a ringing tone for the calling party<br />; g: When<br />> the called party hangs up, exit to execute more commands in the current<br />> context. (new in 1.4)<br />; i: Asterisk will ignore any forwarding (302<br />> Redirect) requests received. Essential for DID usage to prevent fraud.<br />> (new in 1.4) Useful if you are ringing a group of people and one person<br />> has set their phone to forwarded direct to voice
mail on their cell or<br />> something which normally prevents any of the other phones from ringing.<br /><br />> />; R: Indicate ringing<br />> to the calling party when the called party indicates ringing, pass no<br />> audio until answered.<br />; m: Provide Music on Hold to the calling<br />> party until the called channel answers. <br />; L(x[:y][:z]): Limit<br />> the call to 'x' ms, warning when 'y' ms are left, repeated every 'z'<br />> ms)<br />; %timeout% tag is replaced by the calculated timeout<br />> according the credit & destination rate!</p>
<br />>
<p>dialcommand_param = "|60|HRgrL(%timeout%:61000:30000)"</p>
<br />>
<p>; by default (3600000 = 1HOUR MAX CALL)<br /><br />> />dialcommand_param_sipiax_friend = "|60|HRgirL(3600000:61000:30000)"</p>
<br />>
<p>; Define the order to make the outbound call<br />; YES -><br />> SIP/dialedphonenumber@gateway_ip - NO SIP/gateway_ip/dialedphonenumber<br /><br />> />; Both should work exactly the same but i experimented one case when<br />> gateway was supporting dialedphonenumber@gateway_ip <br />; So in case of<br />> trouble, try it out<br />switchdialcommand = NO</p>
<br />>
<p>; failover recursive search - define how many time we want to authorize<br />> the research of the failover trunk when a call fails (value : 0 - 20)<br /><br />> />failover_recursive_limit = 2</p>
<br />>
<p>; For free calls, limit the duration: amount in seconds <br /><br />> />maxtime_tocall_negatif_free_route = 5400</p>
<br />>
<p>; Send a reminder email to the user when they are under<br />> min_credit_2call <br />send_reminder = NO</p>
<br />>
<p>; enable to monitor the call (to record all the conversations)<br />;<br />> value : YES - NO<br />record_call = NO</p>
<br />>
<p>; format of the recorded monitor file <br />monitor_formatfile =<br />> gsm</p>
<br />>
<p>; Force to play the balance to the caller in a predefined currency, to<br />> use the currency set for by the customer leave this field empty<br /><br />> />agi_force_currency =</p>
<br />>
<p>; CURRENCY SECTION<br />; Define all the audio (without file<br />> extensions) that you want to play according to currency (use , to<br />> separate, ie "usd:prepaid-dollar,mxn:pesos,eur:Euro,all:credit")<br /><br />> />currency_association = usd:dollars,mxn:pesos,eur:euros,all:credit</p>
<br />>
<p>; Please enter the file name you want to play when we prompt the<br />> calling party to enter the destination number<br />;<br />> file_conf_enter_destination = prepaid-enter-number-u-calling-1-or-011<br /><br />> />file_conf_enter_destination = prepaid-enter-dest</p>
<br />>
<p>; Please enter the file name you want to play when we prompt the<br />> calling party to choose the prefered language<br />;<br />> file_conf_enter_menulang = prepaid-menulang<br />file_conf_enter_menulang<br />> = prepaid-menulang2</p>
<br />>
<p>; Define if you want to bill the 1st leg on callback even if the call<br />> is not connected to the destination<br /><br />> />callback_bill_1stleg_ifcall_notconnected = YES</p>
<br />>
<p>Â</p>
<br />>
<p><br />Em 27/04/2009 17:38,<br />> <strong><span>pruonckk@pruonckk.org</span></strong> escreveu:</p>
<br />>
<blockquote style="border-left: 2px solid #6868cc; padding-left: 1ex;"><br /><br />verifique o parametro que está<br />> sendo utilizado na discagem pelo a2billing<br />em<br />> /etc/asterisk/a2billing.conf<br /><br />><br />>
<p>Pessoal, tenho um pequeno escritório onde os clientes não<br />> conseguem<br />> transferir as ligações da forma correta.</p>
<br />> <br />><br />>
<p>Tenho o A2B instalado, quando faço uma ligação e vou<br />> transferÃÂ-la<br />> ela simplesmente não vai. Posso apertar *2<br />> (transferir) quantas vezes for<br />> que o cliente do outro lado da<br />> linha escuta o DTMF do *2 mas a ligação<br />> não vai, é<br />> como se o A2B não reconhecesse essa facilidade, o efeito é<br />><br />> o mesmo que apertar qualquer tecla do telefone durante uma chamada.<br />> Só<br />> consigo transferir usando a tecla TRANSFER do meu IP-Fone<br />> para um RAMAL B,<br />> mas mesmo assim a transferência é feita<br />> porém a chamada externa fica<br />> muda. Então faço uma<br />> segunda transferência do RAMAL B para o ramal<br />> original e,<br />> só assim, tudo passa a funcionar normalmente.</p>
<br />> <br />><br />>
<p>Ã</p>
<br />> <br />><br />>
<p>Não sei se me fiz entender, resumindo, apenas após duas<br />><br />> transferências é que consigo trabalhar com a chamada dentro do<br />> *.</p>
<br />> <br />><br />>
<p>No sip_additional.conf já coloquei transfer=yes e tudo continuou<br />> na<br />> mesma.</p>
<br />> <br />><br />>
<p>Alguma luz?</p>
<br />> <br />><br />>
<p>Ã</p>
<br />> <br />><br />>
<p>Grato,</p>
<br />> <br />><br />>
<p>João Queiroz</p>
<br />> <br />> _______________________________________________<br />><br />> Openmoko Freerunner, primeiro telefone open source, disponÃvel no<br />> Brasil<br />> rodando o Android da Google.<br />><br />> http://www.neodroid.com<br />><br />> Compre uma camiseta da<br />> AsteriskBrasil.org!<br />> http://www.voipmania.com.br<br />><br /><br />> />> Acesse o canal IRC de discussão sobre Asterisk em Português<br />> Brasileiro na<br />> rede Freenode.net: #asterisk-br<br />><br />> _______________________________________________<br />> Lista de<br />> discussões AsteriskBrasil.org<br />><br />> AsteriskBrasil@listas.asteriskbrasil.org<br />><br />> http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil<br /><br /><br />> /><br />_______________________________________________<br />Openmoko<br />> Freerunner, primeiro telefone open source, disponÃvel no Brasil roda
ndo o<br />> Android da Google.<br />http://www.neodroid.com<br /><br />Compre uma<br />> camiseta da AsteriskBrasil.org!<br />http://www.voipmania.com.br<br /><br /><br />> />A<br />> cesse o canal IRC de discussão sobre Asterisk em Português Brasileiro<br />> na rede Freenode.net: #asterisk-br<br /><br />> />_______________________________________________<br />Lista de<br />> discussões AsteriskBrasil.org<br /><br />> />AsteriskBrasil@listas.asteriskbrasil.org<br /><br />> />http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil<br /><br />> /><br /></blockquote>
<br />> _______________________________________________<br />> Openmoko Freerunner, primeiro telefone open source, disponível no Brasil<br />> rodando o Android da Google.<br />> http://www.neodroid.com<br />><br />> Compre uma camiseta da AsteriskBrasil.org!<br />> http://www.voipmania.com.br<br />><br />> Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na<br />> rede Freenode.net: #asterisk-br<br />> _______________________________________________<br />> Lista de discussões AsteriskBrasil.org<br />> AsteriskBrasil@listas.asteriskbrasil.org<br />> http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil<br /><br /><br />_______________________________________________<br />Openmoko Freerunner, primeiro telefone open source, disponível no Brasil rodando o Android da Google.<br />http://www.neodroid.com<br /><br />Compre uma camiseta da AsteriskBrasil.org!<br />http://www.voipmania.com.br<br /><br />A
cesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na rede Freenode.net: #asterisk-br<br />_______________________________________________<br />Lista de discussões AsteriskBrasil.org<br />AsteriskBrasil@listas.asteriskbrasil.org<br />http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil<br /><br /></blockquote>