Pruonckk, lhe devo desculpas, realmente o que você postou era o que estava acontecendo.
Muitíssimo obrigado pela ajuda. João Queiroz Em Seg, 2009-04-27 às 18:22 -0300, pruon...@pruonckk.org escreveu: > o transfer voce tem que por o parametro t ou T (ou os dois) , abaixo > descrição > > # t: Allow the called user to transfer the call by hitting the blind xfer > keys (features.conf) > > * If you have set the variable GOTO_ON_TRANSFER then the transferrer > will be sent to the context|exten|pri (you can use ^ to represent | to > avoid escapes), example: SetVar(GOTO_ON_TRANSFER=woohoo^s^1); works > with both t and T > * WARNING: GOTO_ON_TRANSFER does not exist in any version of ASTERISK > and will not! the variable is called GOTO_ON_BLINDXFR see > http://svn.digium.com/view/asterisk?rev=5495&view=rev and > http://bugs.digium.com/view.php?id=4056 for details. THX to the person > who shared the information above! > > # T: Allow the calling user to transfer the call by hitting the blind xfer > keys (features.conf) > > > > > <p>Antes de postar aqui eu dei uma boa googleada e encontrei algo sobre o > > parâmetro "i" no dialcommand porém não obtive sucesso, abaixo segue > > meus parâmetros de discagem. Se tive paciência segue também meu > > a2billing.conf completo.</p> > > <p>dialcommand_param = "|60|HRgrL(%timeout%:61000:30000)"</p> > > <p>; by default (3600000 = 1HOUR MAX CALL)<br > > />dialcommand_param_sipiax_friend = "|60|HRgirL(3600000:61000:30000)"</p> > > <p>; Define the order to make the outbound call<br />; YES -> > > SIP/dialedphonenum...@gateway_ip - NO SIP/gateway_ip/dialedphonenumber<br > > />; Both should work exactly the same but i experimented one case when > > gateway was supporting dialedphonenum...@gateway_ip <br />; So in case of > > trouble, try it out<br />switchdialcommand = NO</p> > > <p> </p> > > <p>Tentei também alterar o contexto do a2billing para o descrito aqui: > > http://forum.asterisk2billing.org/viewtopic.php?f=16&t=3044&start=15 > > também sem sucesso.</p> > > <p> </p> > > <p>Minha instalação roda no trixbox e os ramais foram importados usando > > o bulk_extensions.</p> > > <p> </p> > > <p>Mais alguma idéia?</p> > > <p>Segue meu a2billing.conf.</p> > > <p>Obrigado,</p> > > <p>João Queiroz</p> > > <p>______________________________</p> > > <p> </p> > > <p>;<br />; config file for the A2Billing Callingcard platform<br />;</p> > > <p><br />; Global Database Setup - select the database type and > > authentication as required.</p> > > <p>[database]<br />hostname = localhost<br />port = 5432<br />user = > > a2billinguser<br />password = a2billing<br />dbname = mya2billing<br > > />;dbtype = postgres<br />dbtype = mysql</p> > > <p><br />[global]<br />; len_cardnumber is removed<br />; interval for the > > length of the cardnumber (number of digits), minimum lenght is 4<br />; > > ie: 10-15 (cardnumber authorised 10, 11, 12, 13, 14, 15) ; 10,12,14 > > (cardnumber authorised 10, 12, 14)<br />interval_len_cardnumber = 10</p> > > <p>; Alias-Card length<br />len_aliasnumber = 10</p> > > <p>; Voucher length<br />len_voucher = 10</p> > > <p>;base currency define the default currency that you want to use to > > setup your system (see the currency table to know the currency code)<br > > />base_currency = brl</p> > > <p>; filename of the image that will be display at the top of the invoice > > (if not defined no image will appear ; path to place the image > > templates/default/images/)<br />; the type of file have to be a > > jpeg/jpg<br />invoice_image = asterisk01.jpg</p> > > <p>; DID Billing - amount of day before the end of the monthly reservation > > to bill the customer to for the DID use<br />; if the user dont have > > enough credit he will get an email asking him to refill<br > > />didbilling_daytopay = 5</p> > > <p>;webiste administrator email address<br />admin_email = > > are...@gmail.com</p> > > <p>; MANAGER CONNECTION PARAMETERS<br />manager_host = localhost<br > > />manager_username = a2billinguser<br />manager_secret = a2billing</p> > > <p><br />; CALL-BACK<br />[callback]<br />; When web call-back is enabled > > this is the context to sent the call.<br />context_callback = > > a2billing-callback</p> > > <p>; this is the Extension to redirect the call when the web callback is > > returned<br />extension = 1000</p> > > <p>; this is the number of seconds to wait before initiating the call > > back.<br />sec_wait_before_callback = 10</p> > > <p>;Number of seconds before the call-back can be re-initiated from the > > web page<br />; to prevent repeated and unwanted calls. <br > > />sec_avoid_repeate = 30</p> > > <p>; if the callback doesnt succeed within the value below, then the call > > is deemed to have failed.<br />timeout = 20</p> > > <p>; if we want to manage the answer on the call<br />; Disabling this for > > callback trigger numbers makes it ring not hang up.<br />answer_call = > > yes</p> > > <p><br />; PREDICTIVE DIALER<br />; number of calls an agent will do when > > the call button is clicked<br />nb_predictive_call = 10</p> > > <p>; Number of days to wait before the number becomes available to call > > again.<br />nb_day_wait_before_retry = 1</p> > > <p>; The context to redirect the call for the predictive dialer<br > > />context_preditctivedialer = a2billing-predictivedialer</p> > > <p><br />; When a call is made we need to limit the call duration : amount > > in seconds <br />predictivedialer_maxtime_tocall = 5400</p> > > <p>; set the callerID for the predictive dialer and call-back<br > > />callerid = 123456</p> > > <p>; ID Call Plan to use when you use the all-callback mode, check the ID > > in the "list Call Plan" - WebUI<br />all_callback_tariff = 1</p> > > <p>; Define the group of servers that are going to be used by the > > callback<br />id_server_group = 1</p> > > <p>; Audio intro message when the callback is initiate <br > > />callback_audio_intro = prepaid-callback_intro</p> > > <p><br />; CUSTOMISATION Of THE CUSTOMER INTERFACE<br > > />[webcustomerui]</p> > > <p>; url of the signup page to show up on the sign in page (if empty no > > link will show up)<br />signup_page_url =</p> > > <p>;Enable or disable the payment methods; yes for multi-payment or no for > > single payment method option<br />paymentmethod = no</p> > > <p>;Enable or disable the page which allow customer to modify its personal > > information<br />personalinfo = no</p> > > <p>; Enable display of the payment interface - yes or no<br />customerinfo > > = no</p> > > <p>; Enable display of the sip/iax info - yes or no<br />sipiaxinfo = > > no</p> > > <p>; Enable the Call history - yes or no<br />cdr = yes</p> > > <p>; Enable invoices - yes or no<br />invoice =no</p> > > <p>; Enable the voucher screen - yes or no<br />voucher = no</p> > > <p>; Enable the paypal payment buttons - yes or no<br />paypal = no</p> > > <p>; Allow Speed Dial capabilities - yes or no<br />speeddial = no</p> > > <p>; Enable the DID (Direct Inwards Dialling) interface - yes or no<br > > />did = no</p> > > <p>; Show the ratecards - yes or no<br />ratecard = no</p> > > <p>; Offer simulator option on the customer interface - yes or no<br > > />simulator = yes</p> > > <p>; Enable the callback option on the customer interface - yes or no<br > > />callback = no</p> > > <p>; Enable the predictivedialer option on the customer interface - yes or > > no<br />predictivedialer = no</p> > > <p>; Let users use SIP/IAX Webphone (Options : yes/no)<br />webphone = > > yes</p> > > <p>;IP address or domain name of asterisk server that would be used by the > > web-phone<br />webphoneserver = localhost</p> > > <p>; Let the users add new callerid<br />callerid = no</p> > > <p>; Let the user change the webui password<br />password = yes</p> > > <p>; The total number of callerIDs for CLI Recognition that can be add by > > the customer<br />limit_callerid = 5</p> > > <p>; Email address to send the notification and error report - new DIDs > > assigned will also be emailed.<br />error_email = > > confidenc...@confidencial.com</p> > > <p>; URL for specific return if an error occur after login<br > > />return_url_distant_login =</p> > > <p>; URL for specific return if an error occur after forgetpassword<br > > />return_url_distant_forgetpassword =</p> > > <p><br />;SIP & IAX client configuration information.<br > > />[sip-iax-info]</p> > > <p>;Trunk Name to show in sip/iax info <br />sip_iax_info_trunkname = > > call-labs</p> > > <p>;Allowed Codec, ulaw, gsm, g729<br />; use multi value without spaces : > > "gsm,ulaw,g729"<br />sip_iax_info_allowcodec = g729</p> > > <p>;host information<br />sip_iax_info_host = call-labs.com</p> > > <p>;IAX Additional Parameters<br />iax_additional_parameters = > > "canreinvite = no"</p> > > <p>;SIP Additional Parameters<br />sip_additional_parameters = "trustrpid > > = yes | sendrpid = yes | canreinvite = no"</p> > > <p>[epayment_method]<br />enable = no<br />; eg, http://localhost - > > should not be empty for productive servers<br />http_server = > > "http://www.call-labs.com"<br />; eg, https://localhost - Enter here your > > Secure Server Address, should not be empty for productive servers<br > > />https_server = "http://www.call-labs.com"<br />; Enter your Domain Name > > or IP Address, eg, 26.63.165.200<br />http_cookie_domain = > > 26.63.165.200<br />; Enter your Secure server Domain Name or IP Address, > > eg, 26.63.165.200<br />https_cookie_domain = 26.63.165.200<br />; Enter > > the Physical path of your Application on your server<br />http_cookie_path > > = "/A2BCustomer_UI/"<br />; Enter the Physical path of your Application on > > your Secure Server<br />https_cookie_path = "/A2BCustomer_UI/"<br />; > > Enter the Physical path of your Application on your server<br > > />dir_ws_http_catalog = "/A2BCustomer_UI/"<br />; Enter the Physical path > > of your Application on your Secure Server<br />dir_ws_https_catalog = > > "/A2BCustome > > r_UI/"<br />; secure webserver for checkout procedure?<br />enable_ssl = > > yes</p> > > <p>http_domain = 26.63.165.200</p> > > <p>dir_ws_http = "/~areski/svn/a2billing/payment/A2BCustomer_UI/"</p> > > <p>; maybe try with :<br />; Define here the URL to notify the payment<br > > />; payment_notify_url=...</p> > > <p>;define the different amount of purchase that would be available - 5 > > amount maximum (5:10:15)<br />purchase_amount = 1:2:5:10:20</p> > > <p>; Item name that would be display to the user when he will buy > > credit<br />item_name = "Credit Purchase"</p> > > <p>; Currency for the Credit purchase, only one can be define here<br > > />currency_code = USD</p> > > <p>; Define here the URL of paypal gateway the payment (to test with > > paypal sandbox)<br />paypal_payment_url = > > "https://secure.paypal.com/cgi-bin/webscr"<br />;paypal_payment_url = > > "https://www.sandbox.paypal.com/cgi-bin/webscr"</p> > > <p>; paypal transaction verification url<br />paypal_verify_url = > > "ssl://www.paypal.com"<br />;paypal_verify_url = > > www.sandbox.paypal.com</p> > > <p>; Define here the URL of Authorize gateway <br />authorize_payment_url > > = "https://secure.authorize.net/gateway/transact.dll"<br > > />;authorize_payment_url = > > "https://test.authorize.net/gateway/transact.dll"</p> > > <p>;paypal store name to show in the paypal site when customer will go to > > pay<br />store_name = Asterisk2Billing</p> > > <p>;Transaction Key for security of Epayment Max length of 60 > > Characters.<br />transaction_key = asdf1212fasd121554sd4f5s45sdf</p> > > <p>;Moneybookers secret word<br />moneybookers_secretword = areski<br /> > > <br />; SIGNUP MODULE<br />[signup]<br />; enable the signup module<br > > />enable_signup = 1</p> > > <p>; enable Captcha on the signup module (value : YES or NO)<br > > />enable_captcha = YES</p> > > <p>; amount of credit applied to a new user.<br />credit = 0</p> > > <p>; the list of id of call plans which will be shown in signup.<br > > />callplan_id_list = 1, 2</p> > > <p>; Specify whether the card is created as active or pending<br > > />activated = no</p> > > <p>; Simultaneous or non concurrent access with the card - 0 = INDIVIDUAL > > ACCESS or 1 = SIMULTANEOUS ACCESS<br />simultaccess = 0</p> > > <p>;PREPAID CARD = 0 - POSTPAY CARD = 1<br />typepaid = 0</p> > > <p>; Define credit limit, which is only used for a POSTPAY card. <br > > />creditlimit = 999999999</p> > > <p>; Authorise the recurring service to apply on this card - Yes 1 - No > > 0<br />runservice = 0</p> > > <p>; Enable the expiry of the card - Yes 1 - No 0<br />enableexpire = > > 0</p> > > <p>; Expiry Date format YYYY-MM-DD HH:MM:SS. For instance, '2004-12-31 > > 00:00:00' <br />expirationdate =</p> > > <p>; The number of days after which the card will expire <br />expiredays > > = 0</p> > > <p>; Create a sip account from signup ( default : yes )<br />sip_account = > > yes</p> > > <p>; Create an iax account from signup ( default : yes )<br />iax_account > > = yes</p> > > <p>; active card after the new signup. if No, the Signup confirmation is > > needed and an email will be sent <br />; to the user with a link for > > activation (need to put the link into the Signup mail template)<br > > />activatedbyuser = no</p> > > <p>; url of the customer interface to display after activation<br > > />urlcustomerinterface = http://localhost/A2BCustomer_UI/</p> > > <p>; Define if you want to reload Asterisk when a SIP / IAX Friend is > > created at signup time<br />reload_asterisk_if_sipiax_created = no</p> > > <p><br />;BACK-UP AND RESTORE<br />; configuration for backup and > > restore<br />[backup]</p> > > <p>; Path to store backup of database<br />backup_path = /tmp</p> > > <p>; path for gzip<br />gzip_exe = /bin/gzip</p> > > <p>; path for gunzip<br />gunzip_exe = /bin/gunzip</p> > > <p>; path for mysqldump<br />mysqldump = /usr/bin/mysqldump</p> > > <p>; path for pg_dump<br />pg_dump = /usr/bin/pg_dump</p> > > <p>; path for mysql<br />mysql = /usr/bin/mysql</p> > > <p>;path for psql<br />psql = /usr/bin/psql</p> > > <p> </p> > > <p>; WEB INTERFACE AND API CONFIGURATION<br />[webui]</p> > > <p>; Path to store the asterisk configuration files SIP & IAX<br > > />buddy_sip_file = /etc/asterisk/additional_a2billing_sip.conf<br > > />buddy_iax_file = /etc/asterisk/additional_a2billing_iax.conf</p> > > <p>; API have a security key to validate the http request, the key has to > > be sent after applying md5 <br />; Valid characters are [a-z,A-Z,0-9]<br > > />api_security_key = Ae87v56zzl34v</p> > > <p>; API to restrict the IP's authorised to make a request. <br />; Define > > The the list of ips separated by ;<br />api_ip_auth = 127.0.0.1</p> > > <p>; Administative Email(not used yet)<br />email_admin = > > confidenc...@confidencial.com</p> > > <p>; MOH (Music on Hold) base directory<br />dir_store_mohmp3 = > > /var/lib/asterisk/mohmp3</p> > > <p>; Number of MOH classes you have created in musiconhold.conf : acc_1, > > acc_2... acc_10 class etc...<br />num_musiconhold_class = 10</p> > > <p>; Display the help section inside the admin interface (YES - NO)<br > > />show_help = YES</p> > > <p>; File Upload parameters<br />; PLEASE CHECK ALSO THE VALUE IN YOUR > > PHP.INI THE LIMIT IS 2MG BY DEFAULT<br />my_max_file_size_import = 1024000 > > ; 1 MG</p> > > <p>; Not used yet, The goal is to upload files and use them in the IVR<br > > />dir_store_audio = /var/lib/asterisk/sounds/a2billing</p> > > <p>; upload maximum file size<br />my_max_file_size_audio=3072000 ; in > > bytes</p> > > <p>; File type extensions permitted to be uploaded such as "gsm, mp3, wav" > > (separated by ,)<br />file_ext_allow = gsm, mp3, wav</p> > > <p>; File type extensions permitted to be uploaded for the musiconhold > > such as "gsm, mp3, wav" (separate by ,)<br />file_ext_allow_musiconhold = > > mp3</p> > > <p><br />; RECORDED CONVERSATIONS</p> > > <p>; Enable link on the CDR viewer to the recordings. (YES - NO)<br > > />link_audio_file = yes</p> > > <p><br />; Path to link the recorded monitor files<br />monitor_path = > > /var/spool/asterisk/monitor<br />; grant access to apache user on read > > mode for the directory :> chmod 755 /var/spool/asterisk/monitor/</p> > > <p>; FORMAT OF THE RECORDED MONITOR FILE <br />monitor_formatfile = > > gsm</p> > > <p>; Display the icon in the invoice<br />show_icon_invoice = YES</p> > > <p>;CURRENCY AND GENERAL SETTINGS</p> > > <p>; Display the top frame (useful if you want to save space on your > > little tiny screen )<br />show_top_frame = NO</p> > > <p>; Allow the customer to chose the most appropriate currency ("all" can > > be used)<br />currency_choose = usd, eur, cad, hkd</p> > > <p>; field to export in csv format from cc_card table<br > > />card_export_field_list = id, username, useralias, lastname, credit, > > tariff, activated, language, inuse, currency, sip_buddy, iax_buddy, > > nbused, mac_addr</p> > > <p>; field to export in csv format from cc_voucher table<br > > />voucher_export_field_list = id, voucher, credit, tag, activated, > > usedcardnumber, usedate, currency</p> > > <p>; Advanced mode - Display additional configuration options on the > > ratecard (progressive rates, musiconhold, ...)<br />advanced_mode = NO</p> > > <p>; Delete the SIP/IAX Friend & callerid when a card is deleted<br > > />delete_fk_card = yes</p> > > <p><br />; This section is basically used when we create a new friend <br > > />; when you create a SIP IAX friend for a card the following parameters > > will define the default value for the peer creation<br />[peer_friend]<br > > />; Refer to sip.conf & iax.conf documentation for the meaning of > > those parameters<br />; sip.conf -> > > http://www.voip-info.org/wiki-Asterisk+config+sip.conf<br />; iax.conf > > -> http://www.voip-info.org/wiki-Asterisk+config+iax.conf<br />type = > > friend<br />allow = ulaw,alaw,gsm,g729<br />context = a2billing<br />; use > > "no" or "yes" with quote otherwise the value will be converted to 1 or > > 0<br />nat = "yes"<br />amaflag = billing<br />; use "no" or "yes" with > > quote otherwise the value will be converted to 1 or 0<br />qualify = > > "yes"<br />host = dynamic<br />dtmfmode = RFC2833</p> > > <p><br />[log-files]<br />; To disable application logging, remove/comment > > the log file name aside service</p> > > <p>; cront - recurring process <br />cront_alarm = > > /tmp/cront_a2b_alarm.log<br />cront_autorefill = > > /tmp/cront_a2b_autorefill.log<br />cront_batch_process = > > /tmp/cront_a2b_batch_process.log<br />cront_bill_diduse = > > /tmp/cront_a2b_bill_diduse.log<br />cront_subscriptionfee = > > /tmp/cront_a2b_subscription_fee.log<br />cront_currency_update = > > /tmp/cront_a2b_currency_update.log<br />cront_invoice = > > /tmp/cront_a2b_invoice.log<br />cront_check_account = > > /tmp/cront_a2b_check_account.log</p> > > <p>; paypal log file, to log all the transaction & error<br />paypal = > > /tmp/a2billing_paypal.log</p> > > <p>; epayment log file, to log all the transaction & error<br > > />epayment = /tmp/a2billing_epayment.log</p> > > <p>; Log file to store the ecommerce API requests<br />api_ecommerce = > > /tmp/api_ecommerce_request.log</p> > > <p>; Log file to store the CallBack API requests<br />api_callback = > > /tmp/api_w<br />callback_request.log</p> > > <p>; File to log<br />agi = /tmp/a2billing_agi.log</p> > > <p> </p> > > <p>; configuration for the AGI, different configuration can be defined, ie > > "agi-conf1", "agi-conf2", etc...<br />; the groupid parameter will define > > which process_sections to use. Usage : DeadAGI(a2billing.php|%groupid%)<br > > />; by default agi-conf1 is used<br />[agi-conf1]</p> > > <p>; the debug level<br />; 0=none, 1=low, 2=normal, 3=all<br />debug = > > 1</p> > > <p>; Asterisk Version Information<br />; 1_1,1_2,1_4 By Default it will > > take 1_2 or higher<br />asterisk_version = 1_2</p> > > <p>; Manage the answer on the call<br />answer_call = YES</p> > > <p>; Play audio - this will disable all stream file but not the Get Data > > <br />; for wholesale ensure that the authentication works and than > > number_try = 1<br />play_audio = YES</p> > > <p>; play the goodbye message when the user has finished.<br />say_goodbye > > = NO</p> > > <p>; enable the menu to choose the language<br />; press 1 for English, > > pulsa 2 para el español, Pressez 3 pour Français<br />play_menulanguage > > = NO</p> > > <p><br />; force the use of a language, if you dont want to use it leave > > the option empty<br />; Values : ES, EN, FR, etc... (according to the > > audio you have installed)<br />force_language = BR</p> > > <p>; Introduction prompt : to specify an additional prompt to play at the > > beginning of the application<br />intro_prompt =</p> > > <p>; Minimum amount of credit to use the application<br />min_credit_2call > > = 0</p> > > <p>; this is the minimum duration in seconds of a call in order to be > > billed<br />; any call with a length less than min_duration_2bill will > > have a 0 cost<br />; useful not to charge callers for system errors when a > > call was answered but it actually didn't connect<br />min_duration_2bill = > > 0</p> > > <p>; if user doesn't have enough credit to call a destination, prompt him > > to enter another cardnumber<br />notenoughcredit_cardnumber = YES</p> > > <p>; if notenoughcredit_cardnumber = YES then assign the CallerID to > > the > > new cardnumber<br />notenoughcredit_assign_newcardnumber_cid = NO</p> > > <p><br />; if YES it will use the DNID and try to dial out, without asking > > for the phonenumber to call<br />; value : YES, NO<br />use_dnid = YES</p> > > <p>; list the dnid on which you want to avoid the use of the previous > > option "use_dnid"<br />no_auth_dnid = 2400,2300</p> > > <p>; number of times the user can dial different number<br />number_try = > > 3</p> > > <p>; this will force to select a specific call plan by the Rate Engine<br > > />force_callplan_id =</p> > > <p>; Play the balance to the user after the authentication (values : yes - > > no)<br />say_balance_after_auth = NO</p> > > <p>; Play the balance to the user after the call (values : yes - no)<br > > />say_balance_after_call = NO</p> > > <p>; Play the initial cost of the route (values : yes - no)<br > > />say_rateinitial = NO</p> > > <p>; Play the amount of time that the user can call (values : yes - no)<br > > />say_timetocall = NO</p> > > <p><br />; enable the setup of the callerID number before the outbound is > > made, by default the user callerID value will be use<br />auto_setcallerid > > = YES</p> > > <p>; If auto_setcallerid is enabled, the value of force_callerid will be > > set as CallerID<br />force_callerid =</p> > > <p>; If force_callerid is not set, then the following option ensures that > > CID is set to one of the card's configured caller IDs or blank if none > > available.<br />; NO - disable this feature, caller ID can be anything.<br > > />; CID - Caller ID must be one of the customers caller IDs<br />; DID - > > Caller ID must be one of the customers DID nos.<br />; BOTH - Caller ID > > must be one of the above two items.<br />cid_sanitize = NO</p> > > <p><br />; enable the callerid authentication<br />; if this option is > > active the CC system will check the CID of caller <br />cid_enable = > > NO</p> > > <p>; if the CID does not exist, then the caller will be prompt to enter > > his cardnumber<br />cid_askpincode_ifnot_callerid = YES</p> > > <p>; if the callerID authentication is enable and the authentication fails > > then the user will be prompt to enter his cardnumber<br />; this option > > will bound the cardnumber entered to the current callerID so that next > > call will be directly authenticate<br />cid_auto_assign_card_to_cid = > > NO</p> > > <p>; if the callerID is captured on a2billing, this option will create > > automatically a new card and add the callerID to it <br > > />cid_auto_create_card = NO</p> > > <p>; set the length of the card that will be auto create (ie, 10)<br > > />cid_auto_create_card_len = 10</p> > > <p>; If cid_auto_create_card has been set to YES, the following options > > will define with which configuration we will create the card<br />;<br />; > > billing type of the new card<br />; ( value : POSTPAY or PREPAY)<br > > />cid_auto_create_card_typepaid = POSTPAY</p> > > <p>; amount of credit of the new card<br />cid_auto_create_card_credit = > > 0</p> > > <p>; if postpay, define the credit limit for the card<br > > />cid_auto_create_card_credit_limit = 1000</p> > > <p>; the tariffgroup to use for the new card (this is the ID that you can > > find on the admin web interface)<br />cid_auto_create_card_tariffgroup = > > 6</p> > > <p>; to check callerID over the cardnumber authentication (to guard > > against spoofing)<br />callerid_authentication_over_cardnumber = NO</p> > > <p>; enable the option to call sip/iax friend for free (values : YES - > > NO)<br />sip_iax_friends = NO</p> > > <p>; if SIP_IAX_FRIENDS is active, you can define a prefix for the dialed > > digits to call a pstn number<br />; values : number<br > > />sip_iax_pstn_direct_call_prefix = 555</p> > > <p>; this will enable a prompt to enter your destination number.<br />; if > > number start by sip_iax_pstn_direct_call_prefix we do directly a sip iax > > call, if not we do a normal call<br />sip_iax_pstn_direct_call = NO</p> > > <p>; enable the option to refill card with voucher in IVR (values : YES - > > NO)<br />ivr_voucher = NO</p> > > <p>; if ivr_voucher is active, you can define a prefix for the voucher > > number to refill your card<br />; values : number - don't forget to change > > prepaid-refill_card_with_voucher audio accordingly<br />ivr_voucher_prefix > > = 8</p> > > <p>; When the user credit are below the minimum credit to call > > min_credit<br />; jump directly to the voucher IVR menu (values: YES - > > NO)<br />jump_voucher_if_min_credit = NO</p> > > <p>; Extracharge DIDs, multiple numbers and fees must be separated by > > comma<br />; extracharge_did = 1800XXXXXXX,1888XXXXXXX<br > > />extracharge_did = <br />;extracharge_fee = 0.02,0.03<br > > />extracharge_fee = <br />;extracharge_buyfee = 0.015,0.025<br > > />extracharge_buyfee =</p> > > <p>; List the prefixes that will be stripped off if the call plan requires > > it<br />international_prefixes = 011,00,09</p> > > <p>; More information about the Dial : > > http://voip-info.org/wiki-Asterisk+cmd+dial<br />; 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.<br />; H: Allow the caller to hang up by dialing * > > <br />; r: Generate a ringing tone for the calling party<br />; g: When > > the called party hangs up, exit to execute more commands in the current > > context. (new in 1.4)<br />; i: Asterisk will ignore any forwarding > > (302 > > Redirect) requests received. Essential for DID usage to prevent fraud. > > (new in 1.4) Useful if you are ringing a group of people and one person > > has set their phone to forwarded direct to voicemail on their cell or > > something which normally prevents any of the other phones from ringing.<br > > />; R: Indicate ringing > > to the calling party when the called party indicates ringing, pass no > > audio until answered.<br />; m: Provide Music on Hold to the calling > > party until the called channel answers. <br />; > > L(x[:y][:z]): Limit > > the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' > > ms)<br />; %timeout% tag is replaced by the > > calculated timeout > > according the credit & destination rate!</p> > > <p>dialcommand_param = "|60|HRgrL(%timeout%:61000:30000)"</p> > > <p>; by default (3600000 = 1HOUR MAX CALL)<br > > />dialcommand_param_sipiax_friend = "|60|HRgirL(3600000:61000:30000)"</p> > > <p>; Define the order to make the outbound call<br />; YES -> > > SIP/dialedphonenum...@gateway_ip - NO SIP/gateway_ip/dialedphonenumber<br > > />; Both should work exactly the same but i experimented one case when > > gateway was supporting dialedphonenum...@gateway_ip <br />; So in case of > > trouble, try it out<br />switchdialcommand = NO</p> > > <p>; failover recursive search - define how many time we want to authorize > > the research of the failover trunk when a call fails (value : 0 - 20)<br > > />failover_recursive_limit = 2</p> > > <p>; For free calls, limit the duration: amount in seconds <br > > />maxtime_tocall_negatif_free_route = 5400</p> > > <p>; Send a reminder email to the user when they are under > > min_credit_2call <br />send_reminder = NO</p> > > <p>; enable to monitor the call (to record all the conversations)<br />; > > value : YES - NO<br />record_call = NO</p> > > <p>; format of the recorded monitor file <br />monitor_formatfile = > > gsm</p> > > <p>; 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<br > > />agi_force_currency =</p> > > <p>; CURRENCY SECTION<br />; 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")<br > > />currency_association = usd:dollars,mxn:pesos,eur:euros,all:credit</p> > > <p>; Please enter the file name you want to play when we prompt the > > calling party to enter the destination number<br />; > > file_conf_enter_destination = prepaid-enter-number-u-calling-1-or-011<br > > />file_conf_enter_destination = prepaid-enter-dest</p> > > <p>; Please enter the file name you want to play when we prompt the > > calling party to choose the prefered language<br />; > > file_conf_enter_menulang = prepaid-menulang<br />file_conf_enter_menulang > > = prepaid-menulang2</p> > > <p>; Define if you want to bill the 1st leg on callback even if the call > > is not connected to the destination<br > > />callback_bill_1stleg_ifcall_notconnected = YES</p> > > <p> </p> > > <p><br />Em 27/04/2009 17:38, > > <strong><span>pruon...@pruonckk.org</span></strong> escreveu:</p> > > <blockquote style="border-left: 2px solid #6868cc; margin: 0pt 0pt 0pt > > 0.8ex; padding-left: 1ex;"><br /><br />verifique o parametro que está > > sendo utilizado na discagem pelo a2billing<br />em > > /etc/asterisk/a2billing.conf<br /><br />> > > <p>Pessoal, tenho um pequeno escritório onde os clientes não > > conseguem<br />> transferir as ligações da forma correta.</p> > > <br />> > > <p>Tenho o A2B instalado, quando faço uma ligação e vou > > transferÃÂ-la<br />> ela simplesmente não vai. Posso apertar *2 > > (transferir) quantas vezes for<br />> que o cliente do outro lado da > > linha escuta o DTMF do *2 mas a ligação<br />> não vai, é > > como se o A2B não reconhecesse essa facilidade, o efeito é<br />> > > o mesmo que apertar qualquer tecla do telefone durante uma chamada. > > Só<br />> consigo transferir usando a tecla TRANSFER do meu IP-Fone > > para um RAMAL B,<br />> mas mesmo assim a transferência é feita > > porém a chamada externa fica<br />> muda. Então faço uma > > segunda transferência do RAMAL B para o ramal<br />> original e, > > só assim, tudo passa a funcionar normalmente.</p> > > <br />> > > <p>Ã</p> > > <br />> > > <p>Não sei se me fiz entender, resumindo, apenas após duas<br />> > > transferências é que consigo trabalhar com a chamada dentro do > > *.</p> > > <br />> > > <p>No sip_additional.conf já coloquei transfer=yes e tudo continuou > > na<br />> mesma.</p> > > <br />> > > <p>Alguma luz?</p> > > <br />> > > <p>Ã</p> > > <br />> > > <p>Grato,</p> > > <br />> > > <p>João Queiroz</p> > > <br />> _______________________________________________<br />> > > Openmoko Freerunner, primeiro telefone open source, disponÃvel 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, disponÃvel 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> > > _______________________________________________ > > Openmoko Freerunner, primeiro telefone open source, disponível no Brasil > > rodando o Android da Google. > > http://www.neodroid.com > > > > Compre uma camiseta da AsteriskBrasil.org! > > http://www.voipmania.com.br > > > > Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na > > rede Freenode.net: #asterisk-br > > _______________________________________________ > > Lista de discussões AsteriskBrasil.org > > AsteriskBrasil@listas.asteriskbrasil.org > > http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil > > > _______________________________________________ > Openmoko Freerunner, primeiro telefone open source, disponível no Brasil > rodando o Android da Google. > http://www.neodroid.com > > Compre uma camiseta da AsteriskBrasil.org! > http://www.voipmania.com.br > > Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na > rede Freenode.net: #asterisk-br > _______________________________________________ > Lista de discussões AsteriskBrasil.org > AsteriskBrasil@listas.asteriskbrasil.org > http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil _______________________________________________ Openmoko Freerunner, primeiro telefone open source, disponível no Brasil rodando o Android da Google. http://www.neodroid.com Compre uma camiseta da AsteriskBrasil.org! http://www.voipmania.com.br Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na rede Freenode.net: #asterisk-br _______________________________________________ Lista de discussões AsteriskBrasil.org AsteriskBrasil@listas.asteriskbrasil.org http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil