provided by Star2Billing S.L.

Hi,
I'm sure someone has probably encounter this issue before )

After i enter my pin and it give me the balance and i enter a phone number it says
"the number you have dial is currently unavailable"

I believe the message is because a2billing don't know how to go out my trunk.

What exactly i need to do in trixbox2.8 or a2billing to get this working?

Thanks in advance for the help.

You need to check you have a rate for the destination, and a working trunk. The debug log will help.

hi Jroper,
I finaly norrow it down that its not connecting/syncing up with what i have configure in my rate plan.
To test I delete the trunk configuration and i was still getting the same message. So its not at the trunk but before the trunk

It should check the rate base on the number im dailing and tell me how many minutes i have left. So since Im not getting that information i know it has to do something with the rate plan.

What log should i be looking? because the asterisk -r -vvvv is not giving me any usefull info.

Please help. I have spend a lot of time on this and I'm almost there :0) I dont want to give up now.

1. Test your trunk outside of A2Billing, directly in Asterisk, so you know it's working.
2. Use the rate card simulator to check there is a rate for your destination.
3. In the agi-conf, turn the verbosity up to 4 and watch the asterisk CLI

Joe

Simulator found a rate for your destination
DESTINATION :#1
a CallTime available 40:00 Minutes
Destination Number United States
a Cost per minute 0.05000

So the simulator is working and i tested the trunk and i can make a call out. However I still getting the error message when using a2billing.

I'm out of ideas now (

Ok...I log in to a2billing through the customer side and use the simulator and it work.

Now when i dialed again and put the pin and then the number it ATLEAST NOW tell me how many minutes I have left. Finally )

BUT after it tells me the number of minutes is giving me the same message.

Let me double check my trunk.

Ok now Im getting the following

- AGI Script Executing Application: (DIAL) Options: (SIP/betateckCWU/17324912578|60|HRrL(2400000:61000:30000))
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
-- Called betateckCWU/17324912578|60|HRrL(2400000:61000:30000)
-- Got SIP response 484 "Address Incomplete" back from 38.99.70.232
== Everyone is busy/congested at this time (1:0/0/1)


trixbox1*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
KelvinCWU/XXXXXX 38.99.70.232 N 5060 OK (97 ms)


I read that this message 484 "address incomplete" means "Provider is expecting call will special prefix or he only allows calls to specified destinations, not all over the world. (For example: calls to US only). "

But i can dial the number fine when the trunk is isolated from the a2billing

what else can it be?

In which format wants to have your provider the numbers?

0017324912578

or only

17324912578

Cheers
Skiller

the provider is expecting 17324912578

This is when I DONT use the a2billing just directly from trixbox to the trunk provider

-- Executing [s@macro-dialout-trunk:19] Dial("SIP/300-09c80350", "SIP/kelcwu/18482485811,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
-- Called kelcwu/18482485812
-- SIP/kelcwu-09cf59c8 is making progress passing it to SIP/300-09c80350

But when i use the a2billing

- AGI Script Executing Application: (DIAL) Options: (SIP/KelCWU/18482485811|60|HRrL(2400000:61000:30000))
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
-- Called KelCWU/18482485811|60|HRrL(2400000:61000:30000)
-- Got SIP response 484 "Address Incomplete" back from 38.99.70.232
== Everyone is busy/congested at this time (1:0/0/1)
-- Playing 'prepaid-dest-unreachable' (escape_digits=#) (sample_offset 0)

Hi Betateck,

do you have this prefix in Rates > Rates?

184

What about Sip show peers?

Is this Provider registered"kelcwu"?

Cheers
Skiller

yes i have the rate the the trunk is connected fine.
I contacted my provider and recommended i replace the | with a , in a2billing.conf

I did the change but for some reason i still see the |

18482485811|60|HRrL

Is there any other place i should be looking into.

thanks

Dont replace this.

Did you make a

sip show peers

Cheers

do you know why when i have the 1 configure under a2billing-custom exten => _X.,n,DeadAGI(a2billing.php|1)

the call hangs up?

But when i dont have it the system as me for my pin number and the other good stuff.

[custom-a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,DeadAGI(a2billing.php|1)
exten => _X.,n,Hangup

Sip show peers output:
KelCWU/180 38.99.70.232 N 5060 OK (95 ms)

when i have the 1 after the a2billing.php under the custom-a2billing

-- Executing [2532432588@from-pstn:6] Goto("SIP/66.54.140.46-b78b88f0", "cus
tom-a2billing,2532432588,1") in new stack
-- Goto (custom-a2billing,2532432588,1)
-- Executing [2532432588@custom-a2billing:1] Answer("SIP/66.53.14.46-b78b88
f0", "") in new stack
-- Executing [2532432588@custom-a2billing:2] Wait("SIP/66.53.14.46-b78b88f0
", "1") in new stack
-- Executing [2532432588@custom-a2billing:3] DeadAGI("SIP/66.53.14.46-b78b8
8f0", "a2billing.php|1") in new stack
-- Executing [2532432588@custom-a2billing:4] Hangup("SIP/66.53.14.46-b78b88
f0", "") in new stack

Hi,
at this point looks like a problem I had ... and i didn't solve ...
have a look at the full log and see... you may not have the right codec.
cheers.