provided by Star2Billing S.L.

Hi All,
I am a newbie here so apologies if I am missing the obvious...

I am trying to setup an outbound SIP trunk in A2billing but I keep on getting the message "The number you have dialled is currently unavailable"

My Setup is as follows
-Inbound IAX2 Trunk
-Outbound SIP Trunk (TestTrunk)
-AsteriskNow Extensions

Software
-A2Billing Version 1.7.1
-Asterisk 1.4.24

I have set up the Outbound SIP trunk in AsteriskNow and it works fine through the AsteriskNow extensions .

On A2Billing, I have defined the Outbound SIP trunk using the Trunk name (and not IP). I connect through Inbound IAX2 trunk and do hear the balance etc but when I try to dial out, the system gives me the following error

"The number you have dialled is currently unavailable".


This is what I see on the CLI - Debug


[color=#000080] a2billing.php: file:Class.RateEngine.php - line:1234 - uniqueid:1276248227.57 - app_callingcard: Dialing 'SIP/TestTrunk/92718500000|60|HRrL(8520000:61000:30000)' with timeout of '8520'.
a2billing.php:
a2billing.php: file:Class.RateEngine.php - line:1253 - uniqueid:1276248227.57 - app_callingcard: CIDGROUPID='-1' OUTBOUND CID SELECTED IS '0'.
a2billing.php: file:Class.RateEngine.php - line:1148 - uniqueid:1276248227.57 - [TRUNK STATUS UPDATE : UPDATE cc_trunk SET inuse=inuse+1 WHERE id_trunk='3']
-- AGI Script Executing Application: (DIAL) Options: (SIP/TestTrunk/92718500000|60|HRrL(8520000:61000:30000))
-- Limit Data for this call:
> timelimit = 8520000
> play_warning = 61000
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000
> start_sound = (null)
> warning_sound = timeleft
> end_sound = (null)
Audio is at 94.193.Y.Y port 60170
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 64.79.X.X:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 94.193.Y.Y:5060;branch=z9hG4bK7ab74fcb;rport
From: "07813000000" <sip:[email protected]>;tag=as089b8a66
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Jun 2010 09:24:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 3356 3356 IN IP4 94.193.Y.Y
s=session
c=IN IP4 94.193.Y.Y
t=0 0
m=audio 60170 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called TestTrunk/92718500000
localhost*CLI>
<--- SIP read from 64.79.X.X:5060 --->
SIP/2.0 404 Not Found
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.40:5060;branch=z9hG4bK7ab74fcb
From: "07813000000" <sip:[email protected]>;tag=as089b8a66
Call-ID: [email protected]
To: <sip:[email protected]>;tag=1706451008165650659377921
Contact: <sip:64.79.X.X:5060;transport=udp>
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 64.79.X.X:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 94.193.Y.Y:5060;branch=z9hG4bK7ab74fcb;rport
From: "07813000000" <sip:[email protected]>;tag=as089b8a66
To: <sip:[email protected]>;tag=1706451008165650659377921
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/TestTrunk-09c487e8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
a2billing.php: file:Class.RateEngine.php - line:1262 - uniqueid:1276248227.57 - DIAL SIP/TestTrunk/92718500000|60|HRrL(8520000:61000:30000)
a2billing.php: file:Class.RateEngine.php - line:1148 - uniqueid:1276248227.57 - [TRUNK STATUS UPDATE : UPDATE cc_trunk SET inuse=inuse-1 WHERE id_trunk='3']
[/color]

1. Have you confirmed that you can dial 92718500000 via asterisk alone?
2. is 92718500000 a valid number?

Joe

I have changed the IPS and telephone number for obvious reason but yes, the number works through asterisknow

If you are using Asterisk 1.6, have you changed the asterisk global version to 1_6 in the system settings?

Joe

Yes, already changed the version to 1_4 (I am using AsteriskNow 1.4)

The A2B configuration for asterisk version is there if you care to find it. The default is not 1_6.

It seems your system is behind a NAT. In this case, check to make sure you have set in sip.conf:

externip= (your external IP)
localnet= eg 192.168.0.1/24 or 192.168.0.1/255.255.255.0
realm= (your external IP or domain name)

Not because asterisk is behind a NAT you use a nat= setting. You should use the nat=yes to a peer/friend if you know it is behind a NAT communicating with asterisk.

If your asterisk is on a public IP then you only use the realm= and nat=yes to peers/friends you know are behind a NAT. Otherwise you let asterisk use it's default which is nat=no which attempts nat if detected. If there are no NAT devices anywhere, then you could set a global nat=never so nat is not attempted.

Thanks for response

Sip.conf is configured as you have described above, I have corrected the nat setting but probably neither of the things you have described above are the issue. Remember, the trunk is already working fine from within Asterisk !

Still waiting for help on this one.... Appreciate if one of the forum gurus can help

Tha'ts the message you get when you dial a wrong number or the username/password sent to the proxy is wrong or the user is not registered, sorry.

He same number works fine from asterisk !

And same trunk configuration works fine from asterisk !

Hi

Compare the sip messages when you dial from Asterisk, to when you dial via A2Billing, and see where the difference lies. This may give you a clue to what is going on.

Joe