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 Post subject: A2billing charges for IVR too
PostPosted: Wed Aug 12, 2015 5:56 pm 
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Joined: Wed Aug 12, 2015 5:48 pm
Posts: 4
Hi All,

I have a new implementation of A2billing with Asterisk 12.8.2. I am using Voucher for outgoing calls. Everything is pretty fine so far except charging the calls.

Rates are configured and shows up with correct minutes and credits available. However, the call is being charged from the time I dial in to A2billing IVR.

For instance I am using voucher to dial an external number

Call answered by a2b and prompts for PIN -> I enter PIN -> Asks for number to dial -> I enter the number -> My mobile rings

For the above actions, I may require 30 to 40 seconds of time and I answer the call after 40 seconds from the initiation of the call. Now I am in conversation for 30 seconds.

After the call, I am charged for 40 + 30 = 70 seconds :cry: :( :o instead of actual 30 seconds my mobile was in conversation with SIP extension.

Please help me how to resolve this problem... :)


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 Post subject: Re: A2billing charges for IVR too
PostPosted: Thu Aug 13, 2015 1:44 pm 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Hi

Sounds like your carrier is answering the call, they shouldn't

Joe


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 Post subject: Re: A2billing charges for IVR too
PostPosted: Thu Aug 13, 2015 3:00 pm 
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Joined: Wed Aug 12, 2015 5:48 pm
Posts: 4
No its not carrier because the call wasn't answered by the GSM to VoIP gateway until I enter the destination. Morever, I have noticed in A2billing -> Customer that the IVR is being charged when the prompt is playing. If I didn't make external call, then the charge is refunded or not taken. But if I made a call then the call is being charged for the prompt as well.


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 Post subject: Re: A2billing charges for IVR too
PostPosted: Thu Aug 13, 2015 4:17 pm 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Hi

The point is, A2Billing does not start billing until the call is answered, therefore something upstream of your A2Billing system must be answering the call, as can probably be seen by the logs.

The charge you are seeing taken and refunded is the balance reservation feature inn the AGIConf to alleviate the possibility of negative balance when multiple concurrent calls are being made.

Joe


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 Post subject: Re: A2billing charges for IVR too
PostPosted: Thu Aug 13, 2015 4:36 pm 
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Joined: Wed Aug 12, 2015 5:48 pm
Posts: 4
Joe,

Thanks for your response. I suspect this might be the gateway. Since I am using a GSM to VoIP gateway I suspect gateway might answer my SIP call from asterisk and then try to make the GSM call from the SIM. I am checking the traces and gateway configs. I will let you know if I found something interesting :)


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 Post subject: Re: A2billing charges for IVR too
PostPosted: Thu Aug 13, 2015 6:01 pm 
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Joined: Wed Aug 12, 2015 5:48 pm
Posts: 4
Joe,

I double checked its not a problem with the gateway. I re-emphasize that when the call is answered a2billing charges from the time when I dial into a2billing.php. is there any way

the 'answered time is 66' which is wrong. My mobile shows I am conversation for 40 seconds only.

Log is at https://www.dropbox.com/s/p8qobrh87ddg7 ... g.txt?dl=0

My call started ringing at around

a2billing.php,1: file:Class.RateEngine.php - line:1223 - uniqueid:1439527873.56 - [TRUNK STATUS UPDATE : UPDATE cc_trunk SET inuse = inuse + 1 WHERE id_trunk = '2']
-- AGI Script Executing Application: (DIAL) Options: (SIP/gsm0/9066371511,60,L(4140000:61000:30000))
> Limit Data for this call:
> timelimit = 4140000 ms (4140.000 s)
> play_warning = 61000 ms (61.000 s)
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000 ms (30.000 s)
> start_sound =
> warning_sound = timeleft
> end_sound =
== Using SIP RTP CoS mark 5
-- Called SIP/gsm0/9066371511
-- SIP/gsm0-00000026 is making progress passing it to SIP/6332247668-00000025
> 0x7f99b43a5890 -- Probation passed - setting RTP source address to 192.168.2.242:8000
-- SIP/gsm0-00000026 answered SIP/6332247668-00000025
-- Channel SIP/6332247668-00000025 joined 'simple_bridge' basic-bridge <505c3511-5d1a-460e-bfca-806bbbe6c504>
-- Channel SIP/gsm0-00000026 joined 'simple_bridge' basic-bridge <505c3511-5d1a-460e-bfca-806bbbe6c504>
> Bridge 505c3511-5d1a-460e-bfca-806bbbe6c504: switching from simple_bridge technology to native_rtp
> 0x7f99b43a5890 -- Probation passed - setting RTP source address to 192.168.2.242:8000
[Aug 14 10:21:46] WARNING[15618][C-00000015]: chan_sip.c:7386 sip_write: Can't send 10 type frames with SIP write


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