provided by Star2Billing S.L.

I would like to know how announce balance in during the call work. I have make a2billing as a ivr server for charging the call that come to listen the sound. when the call come and place destination for the ivr which i make as a extension (local/$extension) to play the sound file. but it doesn't has an announcement of the balance. Anyone can explain ?

Check in /var/lib/asterisk/sounds/digits

The balance cannot be found in a single sound file. It is played but using multiple files like:

For 176 ---> 4 files: 1.gsm, hundred.gsm, 70.gsm, 6.gsm

And Asterisk has some applications that can be used in order to play souds stuff. Maybe one of those applications is being used via the Asterisk Manager when playing the balance. In that case, it will not be as obvious as you expect it to be.

ex. The following functions will say something to the connected user.

- SayNumber(digits[,gender])
- SayDigits(digits)
- SayAlpha(string)
- SayUnixTime([unixtime][|[timezone][|format]])

I mean play warning file. i have change the dial command in Class.Rateengine.php to be play stream file. which during the playing sound can be played the warning for the balance. Any suggest.


Thank you

If you are talking about the warning message that comes off lets say 60 seconds before the end of the call that says "You have xx seconds remaining for this call". If you want to modify it to make it say something else, well, you should know that it is also done directly by Asterisk. And more precisely, by the Dial command (read more at http://www.voip-info.org/wiki-Asterisk+cmd+dial).

Here is the L parameter of the Asterisk Dial command:



You will see some configuration lines in a2billing.conf that specify some settings related to the warning message.

Hi .

I try to set one sip account by using ip loopback to register the sip account. which the call come in will dial to that extension to play the sound. but i still cannot warning message. Any suggest.

In sip.com

register => 09:[email protected]

[09]
type=friend
username=2102
secret=2102
host=dynamic
qualify=yes
dtmfmode=rfc2833
context=ivr
canreinvite=no
allow=all
cancallforward=no
nat=yes


For the extension.conf

exten => _X.,1,DeadAGI(ivr.php)
exten => _X.,n,Hangup
exten => 09,1,Answer()
exten => 09,n,Playback(ivrsound/09)
exten => 09,n,Hangup


but I got the message in the below

ivr.php: line:1127 - app_callingcard: Dialing 'SIP/127.0.0.1/09|30|HS(120)' with timeout of '120'.
ivr.php:
ivr.php: [-- Play IVR sound with timeout of '120'---]
ivr.php: [-- Sound file = 'ivrsound/09'---]
-- AGI Script Executing Application: (Dial) Options: (SIP/127.0.0.1/09|30|HS(120))
-- Setting call duration limit to 120 seconds.
-- Called 127.0.0.1/09
-- Got SIP response 482 "Loop Detected" back from 127.0.0.1
-- Now forwarding Zap/18-1 to 'Local/09@ivr' (thanks to SIP/127.0.0.1-d4af)
-- Executing Answer("Local/09@ivr-feb4,2", "") in new stack
-- Executing Playback("Local/09@ivr-feb4,2", "ivrsound/09") in new stack
-- Local/09@ivr-feb4,1 answered Zap/18-1
-- Playing 'ivrsound/09' (language 'en')
ivr.php: line:1254 - -> dialstatus : ANSWER, answered time is 0
ivr.php:


Any comment.

Thank you

When the 60 seconds limit is reached, my system did not announce anything.
Which file is defined the LIMIT_WARNING_FILE variable ?

Code:
# L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left, repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. The following special variables are optional for limit calls: (pasted from app_dial.c)

* LIMIT_PLAYAUDIO_CALLER - yes|no (default yes) - Play sounds to the caller.
* LIMIT_PLAYAUDIO_CALLEE - yes|no - Play sounds to the callee.
* LIMIT_TIMEOUT_FILE - File to play when time is up.
* LIMIT_CONNECT_FILE - File to play when call begins.
* LIMIT_WARNING_FILE - File to play as warning if 'y' is defined. If LIMIT_WARNING_FILE is not defined, then the default behaviour is to announce ("You have [XX minutes] YY seconds").

it's dial command which contain it Class.Rateengine.php