Hi Thorsten Thanks very much, at this point my preference is rfc2833 but I will try some other options.
The system is generating audible tones (that I can hear), although I think the audio is generated by the last sip device in the network so if thats so I don't have any control of it. Probably then I have to go to inband to get some control back, I am not sure what I lose from this, or change upstream provider (although the current provider works from a different system) Cheers Duncan On 12/01/2011, at 11:42 PM, Thorsten Göllner wrote: > As far as I can remember you should take a look at the used codec and this > here: > http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode > > Some codecs do not feel happy with some seetings for dtmfmode. Perhaps you > may comapre these on your 2 boxes. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
