[asterisk-users] DTMF not being heard correctly by far end conference system

[Duncan Turnbull]

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