[Asterisk-Users] Grandstream Early Dial

[Greg Boehnlein]

> >>What happens when you change the configuration of the GS phone to
> >>send DTMF via SIP INFO?
> >>    
> >"Send via SIP, RTP or INLINE AUDIO".
> >
> Make sure you change your "dtmfmode=" in your sip.conf to match the mode 
> set on the phone..

Yes.. that solved it. I added "dtmfmode=info" to sip.conf and set "SIP INFO" as the DTMF type on the phone and all is now well!

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