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Ticket #27 (closed defect: invalid)

Opened 1 year ago

Last modified 1 year ago

SIP Customers

Reported by:	nitesh@vipernetworks.com	Assigned to:	areski
Priority:	major	Milestone:
Component:	General	Version:	Stable 1.3.0 (Yellowjacket)
Keywords:		Cc:

Description

Hello Support,

I got working copy of A2Billing 1.3 on CentOS 4.5 and Asterisk 1.2. I created the Trunk and the Rate Cards according to the documentation. I also created a SIP Customer and using XLite dialer I was able to login and also manage to call out on my SIP Trunk.

Now the question is, how can I remove the IVR prompt? Meaning like every time if I have to call, I can not dial straight away... I have to go thru the IVR prompt saying "You have XXX amount, please enter the phone you wish to call" and then I have to enter the phone... I want to skip that part and just dial straight from the XLite Dialer and let the A2Billing do the rest when I hang-up.

Its some like a post-paid system where you are straight away allowed to call and billed later on.

Is it possible?

Thanks, Nitesh

Change History

(in reply to: ↑ description ) 06/14/07 15:42:06 changed by jroper

status changed from new to closed.
resolution set to invalid.

Replying to nitesh@vipernetworks.com:

Hello Support, I got working copy of A2Billing 1.3 on CentOS 4.5 and Asterisk 1.2. I created the Trunk and the Rate Cards according to the documentation. I also created a SIP Customer and using XLite dialer I was able to login and also manage to call out on my SIP Trunk. Now the question is, how can I remove the IVR prompt? Meaning like every time if I have to call, I can not dial straight away... I have to go thru the IVR prompt saying "You have XXX amount, please enter the phone you wish to call" and then I have to enter the phone... I want to skip that part and just dial straight from the XLite Dialer and let the A2Billing do the rest when I hang-up. Its some like a post-paid system where you are straight away allowed to call and billed later on. Is it possible? Thanks, Nitesh

All mentioned above is configurable through a2billing.conf

06/15/07 10:14:44 changed by nitesh@vipernetworks.com

status changed from closed to reopened.
version set to Stable 1.3 (Yellowjacket).
resolution deleted.

OK, I got everything working... I manage to create a SIP Customer with a real DID number and configured an ATA with the DID number. ATA can login and can make calls out without any issues.

But incoming calls are failing... As soon as the call hits Asterisk, A2Billing script runs and ask for PIN Number... I checked the context for my DID it shows "context=a2billing" and under sip.conf "context=a2billing".

If I change the default context under sip.conf to "context=default", then the calls are failing... meaning I do not get any response back, but on *CLI debug show that its failing to look for the DID number. Well, I know this is due to my DID is in "context=a2billing".

Anyone can suggest how can I fix this... I want to ring my incoming to that ATA which has DID assigned.

Cheers, Nitesh

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