Interesting this. With SIP this usually a mismatch between how DTMF is propagated between endpoints, i.e. RTP, SIPinfo or RFC22833. Whichever method is used both endpoints need to use the same. But you are using IAX so I assume this is not an issue. Perhaps your provider of the incoming calls need to set some variable or other. Maybe its a codec thing, not helpful I know....
Brian > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Sam Bacsa > Sent: 31 March 2004 11:31 > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] VoicePulse Connect & DTMF Tones > > > I am using VoicePulse Connect! to connect to my Asterisk server through > IAX2. > > It appears that when anyone dials my DID and proceeds to dial an > extension when prompted, the DTMF tones are not received by the server > (as in, the server continues to play the incoming message in a loop, not > realizing an extension has been entered). I have even tried > automatically transferring to a VM box and seeing if I can log in -- I > cannot. > > Is there any way to fix this? What are the configuration parameters to > get DTMF to work on incoming calls? > > I am 100% sure that my extensions.conf configuration is in order, so it > is not an Asterisk problem. > > Please Help! > > > Thanks, > Sam Bacsa > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
