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

[Mark Spencer]

Latest CVS should not detect 'f' except in the case of a real fax.

Mark

On Thu, 29 Jan 2004, Brent Franks wrote:

> 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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