We are testing some DTMF-driven applications over VOIP (legacy systems which use fast pulses of standard DTMF tones).
The applications work fine when Digium IAXy's are used - no loss or garbling of DTMF tones. However, when we use SIP modems (such as Sipura 1000's), the DTMF tones are frequently uninterpretable and our applications have to ask for retries. I am under the impression that the IAXy is digitizing DTMF tones and sending just the "pure" data, rather than the audio representation, and that this explains why the IAXY's work flawlessly in this application. I am also under the impression that SIP modems should also support a mode like this.. We have tried: dtmfmode=rfc2833 in "sip.conf", and we have also tried turning on "DTMF Tx:" to "AVT" on the Sipura, but this does not affect reliability at all. So my question is: 1) Are we doing anything wrong, or is there something more we should be doing, to enable DTMF translation (ala rfc2833) in Asterisk and/or our SIP modems? 2) Is there any kind of debugging mode in Asterisk which we can turn on, which will show once and for all whether or not we really have successfully enabled rfc2833? We are using Asterisk 1.0.3, by the way. Thank you very much in advance! Brent _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
