In article <[EMAIL PROTECTED]>, John covici <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
