asterisk users - Dec 2004 - Problem with TDM400P and cidstart=polarity

I'm testing a TDM400P with FXO module to receive incoming calls from an 
analogue line and send it to a SIP device.
To recieve callerid, I need to use cidsignalling=dtmf and cidstart=polarity.
The problem is that when a call is finished, the TDM400P seems to require 
about 20 seconds to prepare for the next incoming call. If a new call comes 
in within 20 seconds after the previous call was hungup, the TDM400P answers 
with a modem carrier, sounding like you're calling a modem pool..!  The 
caller hangs up and retries the call, and the next time everything is OK. 
If the second call comes in later than 20 seconds after the previous call 
was finished, or if I remove the cidstart=polarity (and don't get callerid) 
everything works fine.
I can't see any difference in the Asterisk debug logs worth mentioning... 
Has anyone experienced anything similar..?

On Sat, 11 Dec 2004, Rickard Kristiansson wrote:
> I'm testing a TDM400P with FXO module to receive incoming calls from an
> analogue line and send it to a SIP device.
> To recieve callerid, I need to use cidsignalling=dtmf and
cidstart=polarity.
> The problem is that when a call is finished, the TDM400P seems to require 
> about 20 seconds to prepare for the next incoming call. If a new call comes
> in within 20 seconds after the previous call was hungup, the TDM400P
answers
> with a modem carrier, sounding like you're calling a modem pool..!  The
> caller hangs up and retries the call, and the next time everything is OK. 
> If the second call comes in later than 20 seconds after the previous call 
> was finished, or if I remove the cidstart=polarity (and don't get
callerid)
> everything works fine.
> I can't see any difference in the Asterisk debug logs worth
mentioning...
> Has anyone experienced anything similar..?
It may be that the remote hangup supervision for your line is signalled
via a polarity reversal as well and Asterisk misstankenly thinks that is
the start of the callerid for the next call. The 20 seconds would then be
the time it takes for Asterisk to abandon that phantom call.

A similar issue has been discussed in 
  http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0002847

My comments there describe how the Swedish PSTN works. The signalling
there is built around polarity reversals. At the moment I do not think
Asterisk handles all cases. Since we are isdn based I have not looked
further into it.

Peter