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

[Ricardo Poppi]

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.