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VoIP Billing solution


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 Post subject: Callback - When call ends, Asterisk does not HANGUP
PostPosted: Fri Jan 25, 2008 5:16 pm 
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Joined: Fri Jan 25, 2008 5:09 pm
Posts: 3
Hi,
I am able to make a succesfull call back, The issue when the parties hangup, it is as if Asterisk is unaware of the hangup and no records are written to MySQL. Only when I do a "stop now" on asterisk, does it close the call and record is written to mysql?

In a2billing.conf, I have answer_call=YES. Any suggestion is much appreciated.


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 Post subject:
PostPosted: Fri Jan 25, 2008 6:00 pm 
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Joined: Mon Apr 30, 2007 6:43 am
Posts: 1060
Location: Canada
go in sip.conf and set rtptimeout and rtpholdtimeout to a certain value. It will help you limit the damage.

Basically it will close the channel when no RPT is being detected anymore.

I out boxes, it is set to the following values
Code:
rtptimeout=60
rtpholdtimeout=300


These mean 1 minute of total RTP silence and maximum time to keep someone on hold is 5 minutes.


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 Post subject:
PostPosted: Fri Jan 25, 2008 6:06 pm 
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Thanks. But it would be useful to know WHY Asterisk is NOT aware of call hanging up and fixing the issue at the root. Is anyone else having this issue?


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 Post subject:
PostPosted: Fri Jan 25, 2008 6:21 pm 
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Location: Canada
Your are know asking this question on the right forum.


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 Post subject:
PostPosted: Fri Jan 25, 2008 6:22 pm 
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Joined: Fri Jan 25, 2008 5:09 pm
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Maybe I am am mistaken, but is this not an a2billing issue? Callback has been implemented as part of a2billing?


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 Post subject:
PostPosted: Fri Jan 25, 2008 6:35 pm 
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Your question is ...

dlal wrote:
WHY Asterisk is NOT aware of call hanging up


Personally, I have not clue, but this has nothing to do with A2Billing. An in the VoIP world in general, it is well known that sometimes, channels are not cleared down correctly. That why many VoIP base home phone service provider will put an absolute timeout on all calls to make sure that if the hangup is not detect, the absolution timeout will kill the channel anyway. Usually, it is set to 1h. The down side of this is that you cannot talk on these lines for more than 1 hour. The line will be disconnected and you will have to call again. Not all of them do the absolute timeout. Personally, I think that rtptimeout is much more stylish and it will not wait for up to 1h to free valuable resources not to mention the duration of a messy situation.

Oh, and yeah, definitely, this has nothing to do with a2billing. It may even be your provider. Or it maybe be SIP packet routing issue, it can be anything. But once a2billing connects the 2 calls, all it does is wait for the call to be completed. Asterisk is supposed to report the end of the call to A2Billing, not the other way around.


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