[Asterisk-Users] [Fwd: ASTCC speaks and cut RTP channel => Kind of solution...

[Darren Wiebe]

Have you done any testing to see if it made any difference what type of 
trunk was being used?

Darren Wiebe
darren at aleph-com.net

Ricardo Poppi wrote:

> Hi all.
>
> I´ve found a kind of solution (if we can call it this way...) and Im
> reporting it here to help save some lives.
>
> Editing into astcc.cgi I found where the parameters that set 60 and 30
> seconds warning were and put zeros in its place. The last two
> lots-of-zeros numbers at second line. So the zap trunk code of astcc.cgi
> became like that:
>
> ======================================================================
>      if ($res->{tech} eq "Zap") {
>               $dialstr = "Zap/$res->{path}/$phone|30|HL(" . ($maxtime *
> 60 * 1000) . ":00000:00000)";
>               $res = $AGI->exec("DIAL $dialstr");
>               $answeredtime = $AGI->get_variable("ANSWEREDTIME");
>               $dialstatus = $AGI->get_variable("DIALSTATUS");
>               $callstart = localtime();
>               return $dialstatus;
>       }
> ======================================================================
>
>
> And - at least until now... - everything is working fine. The credit is
> being take from the cards in the right amount and no warnings are being
> given when 60 and 30 seconds left. When credit finishes, the agi script
> just finishes the call.
>
> If somebody has a better way to do that, please let us know.
>
> Rgs, Ricardo Poppi.
>
>
> -------- Mensagem Original --------
> Assunto:     ASTCC speaks and cut RTP channel
> Data:     Fri, 09 Sep 2005 18:09:52 -0300
> De:     Ricardo Poppi <rpoppi77 at gmail.com>
> Para:     asterisk-users at lists.digium.com
>
>
>
> Hi list.
>
> I have a fine running Ser+Asterisk environment and have just installed
> ASTCC. It´s working fine either, including its caller-id authentication
> feature (the one we pass the card-number as CALLERID variable and
> number-to-dial as EXTEN variable).
>
> The issue, a great one, is that when the credit is about one minute to
> end, the ASTCC prompt gets into the call, says that "you have one minute
> left..." and when it was suppose to leave and let the RTP traffic of the
> original call be "reestablished", it never happens. The RTP packets  - I
> could see that at asterisk debug screen - stop running and the call is
> still signaled as active, but no media at all.
>
> This is a serious problem I´m having and, as I could see, I´m not the
> only one. Mr. Chilini reported that around jun 30th this year, as you
> can see bellow: (I just added a comment at this voip-info page to see if
> anyone could give some clues about that)
>
> http://www.voip-info.org/tiki-index.php?page=ASTCCGuide#comments
>
>
> Do anyone here in this list had any situation alike? Do you have any
> clues do help me? (and others because it will be documented, of course).
>
> Thanks in advance,
>
> Ricardo Poppi.
>
>
>
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