I am using asterisk 11.7 for a call balsting project. I am doing it using phpagi and AMI. I am origniating calls using AMI originate function, with application set as an agi file. The call is connecting and I can hear the voice message from the agi file. But I couldn’t track the DTMF input from the callee.
Following are the informations I have. Please help me on this.
Originating the call using AMI with an agi application
$call = $asm->send_request('Originate',
array(
'Channel'=>$_SESSION['auth1'],
'Callerid'=>$_SESSION['phone'],
'Async'=>'yes',
'Timeout'=>30000,
'Application'=>'agi',
'Data' =>"phpagi/firstauth.php,{$_SESSION['memberid']},{$number2},{$_SESSION['phone']},{$blast_file_id}"
)
);
sip.conf
[authuser1]
type=friend
host=dynamic
context=authuser3
disallow=all
allow=alaw
secret=abc122
dtmfmode=inband
[authuser2]
type=friend
host=dynamic
context=authuser
secret=abc123
disallow=all
allow=ulaw
dtmfmode=inband
I have made the call to the first user.
modules.conf
[modules]
autoload=yes
load=pbx_config.so
load=chan_sip.so
load=chan_iax2.so
load=res_rtp_asterisk.so
load=app_hangup.so
load=app_dial.so
load=codec_ulaw.so
load=codec_gsm.so