I think what he was talking about was the fact that Grandstream phones send "f" as a DTMF signal when one hits the "flash" button.

JT


At 11:23 AM -0600 1/29/04, Mark Spencer wrote:
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: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Cristian
 Manoni
 Sent: Thursday, January 29, 2004 11:04 AM
 To: [EMAIL PROTECTED]
 > 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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