[Asterisk-Users] ASTCC speaks and cut RTP channel

[Darren Wiebe]

This is not much help but it is more of a suggestion than I've seen on 
this list on this issue.   Would you be able to try an older release of 
asterisk say 1.00 or right around there.  I really do believe it was 
working back then but I've not played with that exact issue for a long time.

Darren wiebe
darren at aleph-com.net


Ricardo Poppi wrote:

> 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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