We have an application which is primarily DTMF driven (automated on both sides), which we are trying to deploy over VOIP and Asterisk (using some Sipuras and some IAXY's).
We are finding that in around half the cases, the Asterisk server can't decode the DTMF digits from the field office (or at least some of them). Though, when we place voice calls for testing, we can hear eachother quite well. I was wondering if there are any settings in Asterisk and/or in SIP clients such as the Sipuras, which will optimize the connections for DTMF rather than voice? Thank you in advance, Brent _______________________________________________ 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
