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 Post subject: FreePBX & A2Billing CDRs omissions
PostPosted: Sun Jan 09, 2011 9:53 am 
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Joined: Sat Jun 30, 2007 8:29 pm
Posts: 32
Hi,

For few years now we have been using a FreePBX (Trixbox) in combination with A2billing. The FreePBX is used to handle user’s calls management (extensions, trunks); A2billing to handle users’ web portals and billing for out-going calls to PSTN.

Lately we have noticed calls on providers’ CDRs, which were not showing on our A2b records. On further investigation we have found that some percentage of the outgoing calls is missing from the A2b CDRs but we are being billed for, from the provider side. We have found a similar situation with two of our providers, making us believe the problem is in our own system somewhere.

So far we did not succeeded in identifying a pattern to the non-billed calls, side of their existence. When we compare the FreePBX CDR to A2b, for calls dialed to PSTN; we see an inconsistency. The majority of outgoing calls on the FreePBX CDR (95% ~) will be billed on A2b properly, but some percentage will go through and billed only on the provider side.

We are using FreePBX on Trixbox 2.6.2.3, A2billing 1.6.0
Any ideas where or why this might happen, or how to further check the issue?

tnx,

nir


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 Post subject: Re: FreePBX & A2Billing CDRs omissions
PostPosted: Mon Jan 10, 2011 8:55 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Hi

Check you are not allowing transfers by adding the i parameter to the dial command.

Joe


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 Post subject: Re: FreePBX & A2Billing CDRs omissions
PostPosted: Mon Jan 10, 2011 9:19 am 
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Joined: Sat Jun 30, 2007 8:29 pm
Posts: 32
I am not sure what u mean, can you explain a little please?

what is "i parameter"?


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 Post subject: Re: FreePBX & A2Billing CDRs omissions
PostPosted: Mon Jan 10, 2011 9:49 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
http://tinyurl.com/22sphps


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 Post subject: Re: FreePBX & A2Billing CDRs omissions
PostPosted: Mon Jan 10, 2011 10:41 am 
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Joined: Sat Jun 30, 2007 8:29 pm
Posts: 32
you sent me to google "asterisk dial command", what's the point?


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 Post subject: Re: FreePBX & A2Billing CDRs omissions
PostPosted: Mon Jan 10, 2011 10:57 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
The first entry on your google search is:-

http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

And then if you read down there is a list of what the dial parameters do.

You asked what "i" does, and if you read the entry it tells you:-

Quote:
i: Asterisk will ignore any forwarding requests it may receive on this dial attempt. (new in 1.4) Useful if you are ringing a group of people and one person has set their phone to forwarded direct to voicemail on their cell or something which normally prevents any of the other phones from ringing.


You don't want transfers being allowed, because tof the nature of SIP they cannot be billed By A2Billing, because the transfer takes place after the call has been established.


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 Post subject: Re: FreePBX & A2Billing CDRs omissions
PostPosted: Mon Jan 10, 2011 11:28 am 
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Joined: Sat Jun 30, 2007 8:29 pm
Posts: 32
thank you for the information, it might be very (very) important in the way we provide services today.

I want to be positive sure I understand what you have said. I would like to present few scenarios that we use, if I understand you correctly, currently not being billed.

1. forwarding of incoming calls from local DID, to an extension. then forwarding of the call via the FollowMe module to another phone. The A2b supposed to catch and bill the call according to the extension CallerID, forced via extensions_custom.

2. forwarding of incoming calls to a DISA, providing dial-tone and forwarded to a number based on user entry. Same story, we force user extension CLID, based on the incoming call CLID or PIN number entered by user.

does any of the above present a billing issue to A2b?

thanks for the help,


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 Post subject: Re: FreePBX & A2Billing CDRs omissions
PostPosted: Mon Jan 10, 2011 11:48 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Scenario one may be your issue. It depends how the call is moved, either with a SIP Transfer (not billed) or with second call via A2Billing, and the channels bridged together (Billed)

You should be able to very easily see why a sip transfer does not bill the call, provided you understand the SIP protocol, and you understand how asterisk and A2Billing work together.


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 Post subject: Re: FreePBX & A2Billing CDRs omissions
PostPosted: Mon Jan 10, 2011 12:19 pm 
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Joined: Sat Jun 30, 2007 8:29 pm
Posts: 32
yes, now that you pointed us in that direction, i can see the logic.

I am not positive at this point if this is the issue, as we don't bridge the call, we "Dial" it to its final destination.

the Disa process will is like this:
- The DID is directed to server,

- ext-did-custom has an entery specifing user's DID/CID
exten => 45787720/0218282,1,Goto(disa,4,1)

- the user is dialing a PSTN number, starting with "0" and sent to a2b
;a2b
exten => _0.,1,Wait(1)
exten => _0.,n,DeadAGI(a2billing.php|1)
exten => _0.,n,Hangup

- the 0218282 is specified on user A2b account CallerID list.
A2b associate the incoming call based on the CLID and send it out to its destination (theoretically billing the user for the call)

do you think something like the above might be problematic?

thank you,


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 Post subject: Re: FreePBX & A2Billing CDRs omissions
PostPosted: Mon Jan 10, 2011 12:44 pm 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Hi

I see no issues with the DISA entry at all, all those calls will be recorded.

What I mean by bridge, is in fact a dial.

So you have an inbound channel, and an outbound channel, and you join (bridge) the two legs of the channels together to provide end to end connectivity.

Where as a SIP transfer will do a re-invite between your customer's endpoint, and your carrier and change the path of the media stream without your asterisk having to dial another number.

To go back to my original comment, putting the i in the dial command parameter stops the sip transfer from happening.

Joe


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 Post subject: Re: FreePBX & A2Billing CDRs omissions
PostPosted: Mon Jan 10, 2011 1:17 pm 
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Joined: Sat Jun 30, 2007 8:29 pm
Posts: 32
thank you,

we will make some tests, i'll update the thread after.


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