Asterisk +a2billing

Hello

I don’t if this could be the suitable forum for my issue but maybe someone of you met with the same issue as me.

Hello, I would like to ask you the help related to the following scenario that I want to implement it.

I use the following architecture:

phone caller(originator)--------Server SIP Operator1-------|Transit Operator Server(Asterisk 13+A2billing2.2.0)|-------------------Server SIP Operator2------phone called(destination)

When is initiated the call via Operator 1 the call should be handle through A2billing, billed, and forwarded to Operator 2. When the call reach to A2billing a message “Please enter your complete PIN number” is played and the call is closed by a2billing and is not billed and forwarded to Server SIP operator 2.

I tried on others forums but I didn’t get any advice.

My configuration on “Transit Operator Server” is figured below and logs also.

SIP.conf and extensions.conf for "Operator tranzit"
my configuration in SIP.conf is:

[general]
port = 5060
transport = udp
dtmfmode = rfc2833
nat = force_rport,comedia
disallow = all
allow = ulaw
allow = alaw
allow = g729

[operator2]
type = peer
insecure = port,invite
host = 90.91.90.11
sendrpid = yes
trustrpid = yes
qualify = yes
registersip = yes
context = a2billing

[operator1]
type = peer
insecure = port,invite
host = 90.91.90.12
sendrpid = yes
trustrpid = yes
qualify = yes
registersip = yes
context = a2billing

The Extensions.conf configuration is:

[globals]
FEATURES =
DIALOPTIONS =
RINGTIME = 20
FOLLOWMEOPTIONS =
PAGING_HEADER = Intercom
autofallthrough = yes
static = yes
writeprotect = no
clearglobalvars = yes
[default]
include => a2billing

[a2billing]
include => op1_ns1
include => op2_ns2

[op2_ns2]
include => out_op2
[out_op2]
exten = _31[27]XXXXXXXX,1,Agi(a2billing.php,${EXTEN},1)

[op1_ns1]
include => out_op1

[out_op1]
exten = _30[27]XXXXXXXX,1,Agi(a2billing.php,1)

The logs are below:
[16/11/2015 08:56:21]:[file:a2billing.php - line:119 - uniqueid:]:[CallerID:]:[CN:]:[IDCONFIG : 30214344434]
[16/11/2015 08:56:21]:[file:a2billing.php - line:120 - uniqueid:]:[CallerID:]:[CN:]:[MODE : standard]
[16/11/2015 08:56:21]:[file:Class.A2Billing.php - line:755 - uniqueid:1447656981.373]:[CallerID:44458]:[CN:]:[ get_agi_request_parameter = 4445840695 ; SIP/operator1-00000139 ; 1447656981.373 ; ; 30214344434]
[16/11/2015 08:56:21]:[file:a2billing.php - line:167 - uniqueid:1447656981.373]:[CallerID:44458]:[CN:]:[[ANSWER CALL]]
[16/11/2015 08:56:21]:[file:Class.A2Billing.php - line:674 - uniqueid:1447656981.373]:[CallerID:44458]:[CN:]:[[SET CHANNEL(language) en]]
[16/11/2015 08:56:45]:[file:Class.A2Billing.php - line:252 - uniqueid:1447656981.373]:[CallerID:44458]:[CN:-1]:[HANGUP DETECTED!
]
[16/11/2015 08:56:45]:[file:a2billing.php - line:617 - uniqueid:1447656981.373]:[CallerID:44458]:[CN:-1]:[[NO AUTH (CN:, cia_res:-1, CREDIT:)]]
[16/11/2015 08:56:45]:[CallerID:44458]:[CN:-1]:[[exit]]

Any advice/opinion is helpful.

Thank you