[AsteriskBrasil] Transferênciancia de chamadas com o A2B

Terça Abril 28 20:29:19 BRT 2009

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, pruonckk em 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
> > parÃ¢metro "i" no dialcommand porÃ©m nÃ£o obtive sucesso, abaixo segue
> > meus parÃ¢metros de discagem. Se tive paciÃªncia segue tambÃ©m 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 -&gt;
> > SIP/dialedphonenumber em gateway_ip - NO  SIP/gateway_ip/dialedphonenumber<br
> > />; Both should work exactly the same but i experimented one case when
> > gateway was supporting dialedphonenumber em gateway_ip	<br />; So in case of
> > trouble, try it out<br />switchdialcommand = NO</p>
> > <p>Â </p>
> > <p>Tentei tambÃ©m alterar o contexto do a2billing para o descrito aqui:
> > http://forum.asterisk2billing.org/viewtopic.php?f=16&amp;t=3044&amp;start=15
> > tambÃ©m sem sucesso.</p>
> > <p>Â </p>
> > <p>Minha instalaÃ§Ã£o roda no trixbox e os ramais foram importados usando
> > o bulk_extensions.</p>
> > <p>Â </p>
> > <p>Mais alguma idÃ©ia?</p>
> > <p>Segue meu a2billing.conf.</p>
> > <p>Obrigado,</p>
> > <p>JoÃ£o Queiroz</p>
> > <p>______________________________</p>
> > <p>Â </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 =
> > areski em 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 =
> > confidencial em 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 &amp; 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>Â </p>
> > <p>; WEB INTERFACE AND API CONFIGURATION<br />[webui]</p>
> > <p>; Path to store the asterisk configuration files SIP &amp; 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 =
> > confidencial em 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 :&gt;  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 &amp; 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 &amp; iax.conf documentation for the meaning of
> > those parameters<br />; sip.conf -&gt;
> > http://www.voip-info.org/wiki-Asterisk+config+sip.conf<br />; iax.conf
> > -&gt; 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 &amp; error<br />paypal =
> > /tmp/a2billing_paypal.log</p>
> > <p>; epayment log file, to log all the transaction &amp; 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>Â </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 espaÃ±ol, Pressez 3 pour FranÃ§ais<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 &amp; 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 -&gt;
> > SIP/dialedphonenumber em gateway_ip - NO  SIP/gateway_ip/dialedphonenumber<br
> > />; Both should work exactly the same but i experimented one case when
> > gateway was supporting dialedphonenumber em 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>Â </p>
> > <p><br />Em 27/04/2009 17:38,
> > <strong><span>pruonckk em 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 estÃ¡
> > sendo utilizado na discagem pelo a2billing<br />em
> > /etc/asterisk/a2billing.conf<br /><br />&gt;
> > <p>Pessoal, tenho um pequeno escritÃƒÂ³rio onde os clientes nÃƒÂ£o
> > conseguem<br />&gt; transferir as ligaÃƒÂ§ÃƒÂµes da forma correta.</p>
> > <br />&gt;
> > <p>Tenho o A2B instalado, quando faÃƒÂ§o uma ligaÃƒÂ§ÃƒÂ£o e vou
> > transferÃƒÂ-la<br />&gt; ela simplesmente nÃƒÂ£o vai. Posso apertar *2
> > (transferir) quantas vezes for<br />&gt; que o cliente do outro lado da
> > linha escuta o DTMF do *2 mas a ligaÃƒÂ§ÃƒÂ£o<br />&gt; nÃƒÂ£o vai, ÃƒÂ©
> > como se o A2B nÃƒÂ£o reconhecesse essa facilidade, o efeito ÃƒÂ©<br />&gt;
> > o mesmo que apertar qualquer tecla do telefone durante uma chamada.
> > SÃƒÂ³<br />&gt; consigo transferir usando a tecla TRANSFER do meu IP-Fone
> > para um RAMAL B,<br />&gt; mas mesmo assim a transferÃƒÂªncia ÃƒÂ© feita
> > porÃƒÂ©m a chamada externa fica<br />&gt; muda. EntÃƒÂ£o faÃƒÂ§o uma
> > segunda transferÃƒÂªncia do RAMAL B para o ramal<br />&gt; original e,
> > sÃƒÂ³ assim, tudo passa a funcionar normalmente.</p>
> > <br />&gt;
> > <p>Ã‚</p>
> > <br />&gt;
> > <p>NÃƒÂ£o sei se me fiz entender, resumindo, apenas apÃƒÂ³s duas<br />&gt;
> > transferÃƒÂªncias ÃƒÂ© que consigo trabalhar com a chamada dentro do
> > *.</p>
> > <br />&gt;
> > <p>No sip_additional.conf jÃƒÂ¡ coloquei transfer=yes e tudo continuou
> > na<br />&gt; mesma.</p>
> > <br />&gt;
> > <p>Alguma luz?</p>
> > <br />&gt;
> > <p>Ã‚</p>
> > <br />&gt;
> > <p>Grato,</p>
> > <br />&gt;
> > <p>JoÃƒÂ£o Queiroz</p>
> > <br />&gt; _______________________________________________<br />&gt;
> > Openmoko Freerunner, primeiro telefone open source, disponÃvel no
> > Brasil<br />&gt; rodando o Android da Google.<br />&gt;
> > http://www.neodroid.com<br />&gt;<br />&gt; Compre uma camiseta da
> > AsteriskBrasil.org!<br />&gt; http://www.voipmania.com.br<br />&gt;<br
> > />&gt; Acesse o canal IRC de discussÃ£o sobre Asterisk em PortuguÃªs
> > Brasileiro na<br />&gt; rede Freenode.net: #asterisk-br<br />&gt;
> > _______________________________________________<br />&gt; Lista de
> > discussÃµes AsteriskBrasil.org<br />&gt;
> > AsteriskBrasil em listas.asteriskbrasil.org<br />&gt;
> > 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 discussÃ£o sobre Asterisk em PortuguÃªs Brasileiro
> > na rede Freenode.net: #asterisk-br<br
> > />_______________________________________________<br />Lista de
> > discussÃµes AsteriskBrasil.org<br
> > />AsteriskBrasil em 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 em 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 em listas.asteriskbrasil.org
> http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil