[Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for SIP

Cristian Manoni cristian.manoni at nethesis.it
Thu Jan 29 09:04:22 MST 2004

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Hi All
i have continuos error:
Unable to handle DTMF tone 'f' for 'SIP
on the asterisk console.
after this the call hang up.

I have a BGT 101 that make and receive call from the capi channel

Thanks

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