On 27/03/2008, David Nedved <[EMAIL PROTECTED]> wrote: > > So now it seems 1.4.18 is doing the same as 1.2.27 -- working for the > most part but completely ignoring DTMF on incoming SIP calls. >
Perhaps you now need to delve deeper. Capture a UDP trace between your VoIP provider and Asterisk, and another of the same call between Asterisk and a handset. Do this for an ordinary voice call, no IVR menus etc etc. 1) Can you hear the DTMF being sent by the far end by the way? 2) If you use Wireshark to do a VoIP call analysis of the traces, do you receive any DTMF signalling in the RTP stream, or in INFO packets from your VoIP provider? I'm sure there is more... Steve _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
