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

[Brent Franks]

Christian,

You can change channel.c source code to be more forgiving of
unrecognized DTMF tones.

Look for my addition near the bottom of this struct:

	else if (digit == 'f');

Basically I altered channel.c to this:

static int do_senddigit(struct ast_channel *chan, char digit)
{
        int res = -1;

        if (chan->pvt->send_digit)
                res = chan->pvt->send_digit(chan, digit);
        if (!chan->pvt->send_digit || res) {
                /*
                 * Device does not support DTMF tones, lets fake
                 * it by doing our own generation. (PM2002)
                 */
                static const char* dtmf_tones[] = {
                        "!941+1336/50,!0/50",   /* 0 */
                        "!697+1209/50,!0/50",   /* 1 */
                        "!697+1336/50,!0/50",   /* 2 */
                        "!697+1477/50,!0/50",   /* 3 */
                        "!770+1209/50,!0/50",   /* 4 */
                        "!770+1336/50,!0/50",   /* 5 */
                        "!770+1477/50,!0/50",   /* 6 */
                        "!852+1209/50,!0/50",   /* 7 */
                        "!852+1336/50,!0/50",   /* 8 */
                        "!852+1477/50,!0/50",   /* 9 */
                        "!697+1633/50,!0/50",   /* A */
                        "!770+1633/50,!0/50",   /* B */
                        "!852+1633/50,!0/50",   /* C */
                        "!941+1633/50,!0/50",   /* D */
                        "!941+1209/50,!0/50",   /* * */
                        "!941+1477/50,!0/50" }; /* # */
                if (digit >= '0' && digit <='9')
 
ast_playtones_start(chan,0,dtmf_tones[digit-'0'], 0);
                else if (digit >= 'A' && digit <= 'D')
 
ast_playtones_start(chan,0,dtmf_tones[digit-'A'+10], 0);
                else if (digit == '*')
                        ast_playtones_start(chan,0,dtmf_tones[14], 0);
                else if (digit == '#')
                        ast_playtones_start(chan,0,dtmf_tones[15], 0);
                else if (digit == 'f');
                else {
                        /* not handled */
                        ast_log(LOG_WARNING, "Unable to handle DTMF tone
'%c' for '%s'\n", digit, chan->name);
                        return -1;
                }
        }
        return 0;
}

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Cristian
Manoni
Sent: Thursday, January 29, 2004 11:04 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for
SIP

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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