[Asterisk-Users] Grandstream Early Dial

[WipeOut]

Greg Boehnlein wrote:

>On Tue, 30 Dec 2003, Tilghman Lesher wrote:
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>>On Tuesday 30 December 2003 22:16, Greg Boehnlein wrote:
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>>
>>What happens when you change the configuration of the GS phone to
>>send DTMF via SIP INFO?
>>    
>>
>
>I had that set originally. I get the same behavior no matter wether I use 
>"Send via SIP, RTP or INLINE AUDIO".
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Make sure you change your "dtmfmode=" in your sip.conf to match the mode 
set on the phone..

Later..