[Asterisk-Dev] DTMF Error

[Brent Franks]

Brent Franks wrote:
> WARNING[1236268096]: File channel.c, Line 1296 (do_senddigit): Unable
to
> handle DTMF tone 'f' for 'SIP/####-####'
> 

Olle E. Johnson wrote:
>>SIP DTMF only defines A-D, not 'f'
>>http://www.voip-info.org/tiki-index.php?page=SIP+DTMF+Signalling
>>If there an 'f' and who's sending it for what?
>>/O

>>>Brent Franks wrote:
>>>The Polycom Soundpoint IP 500 Phones send a DTMF code of 'f' when the
>>>mic mute button is pressed a second time (e.g. the phone is unmuted).

>>>To resolve this, I simply changed the channel.c file and made it so
that
>>>Asterisk logs the error, but does not return a -1 and disconnect the
>>>call.  I think that returning -1 (thus killing the call) is a bit
>>>extreme for this type of error.

>>>I was able to resolve this two ways.  The way below, I simply created
>>>another else if statement saying: else if (digit == 'f');

>>>Another way to resolve it is to get rid of the return -1;.

>>>- Brent

>>>	if (f && (f->frametype == AST_FRAME_VOICE) &&
>>>	chan->generatordata) {
>>>           /*
>>>            * 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;

For the upcoming release of 0.7.0 should * return -1; for the else
clause above?  It seems that cutting off a call for an unhandled dtmf
tone is a bit servere.  Maybe just simply log the error and send it to
the consolve, but don't return -1?

I would assume that as time progresses and more companies begin to
develop SIP phones, that some of these devices will send odd DTMF tones
that typically aren't handled, like F in my case of the Polycom SIP 500.

Just a thought?

- Brent