provided by Star2Billing S.L.

Hello all

I try to implement the following tranzit architecture:
|||||||||||||||||||||||||||||||||||||||||||||||||||
phone caller--------SIP Operator1------------|Operator tranzit (Asterisk 13+A2billing2.2.0)|-------------------SIP Operator2------phone called ||||||||||||||||||||||||||||||||||||||||||||||||||||

When initiate the call via Operator 1 the call should be handle through A2billing and forward to Operator 2. When the call reach to A2billing a message "Please enter your complete PIN number" and the call is closed without to reach to operator 2 and is not billed in a2billing.

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
;A2Telecom = SIP/A2Telecom
;CID_6001 = 5022
;TESTOP = SIP/TESTOP
;CID_6000 = 5021
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,${EXTEN},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

Eugen

....Has anybody any idea on this topic?

Thank you.

I think you probably need to enter a PIN , before the call can go through.

Seems this is a setup issue, and not carrier problem.