[Asterisk-Users] ASTCC speaks and cut RTP channel
Ricardo Poppi
rpoppi77 at gmail.com
Fri Sep 9 14:09:52 MST 2005
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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