asterisk users - Dec 2006 - problems using the 1.4 version of meetme

Hi.  I am having a strange problem when using the 1.4 version of
asterisk and zaptel.  If I call from a pstn line into the asterisk box
using a phone number which calls the box via sip, then once I am in
the meetme conference nothing happens when I hit the star key -- I
cannot get the user menu.  There is nothing in the logs at all its as
though asterisk never sees the digit at all.  Now if I do the exacct
same procedure but use a phone number which calls my box via a zap
channel -- using a digium card -- it works perfectly.  This problem
seems to be independent of asterisk 1.4 and zaptel 1.4 versions, but I
did  an svn update this morning on both of those.  Now this problem
does NOT occurr with 1.2 at all, I can call my box using sip and the *
is seen by the meetme conference.

This is bugging me and since we are getting near release time, I would
like to be able to go to 1.4 by then.

Thanks a bunch.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici@ccs.covici.com

In article <17803.56162.240480.951218@ccs.covici.com>,
John covici <covici@ccs.covici.com> wrote:
> Hi.  I am having a strange problem when using the 1.4 version of
> asterisk and zaptel.  If I call from a pstn line into the asterisk box
> using a phone number which calls the box via sip, then once I am in
> the meetme conference nothing happens when I hit the star key -- I
> cannot get the user menu.  There is nothing in the logs at all its as
> though asterisk never sees the digit at all.  Now if I do the exacct
> same procedure but use a phone number which calls my box via a zap
> channel -- using a digium card -- it works perfectly.  This problem
> seems to be independent of asterisk 1.4 and zaptel 1.4 versions, but I
> did  an svn update this morning on both of those.  Now this problem
> does NOT occurr with 1.2 at all, I can call my box using sip and the *
> is seen by the meetme conference.
Are the 1.2 and 1.4 on different boxes?
If so, how do their sip.conf files differ?
How is your SIP provider sending DTMF to you? SIP INFO, RFC2833 or inband?

Running the 1.4 setup, can you create a simple dialplan IVR to test
the reception of digits? Or even just a Read, and then echo the results
using a NoOp? The point is to see whether the problem is specific to
Meetme or general within your 1.4 build of Asterisk.

Cheers
Tony
-- 
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org