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