asterisk users - Dec 2008 - Increase DTMF Tone Duration

Hi,

We are running 1.4.22 and have been experiencing problems with certain 
IVRs and DTMF Tone duration.  We would like to be able to increase DTMF 
Tone duration by 50 to 100ms over what the user is pressing on his 
phone.  We have a PRI test circuit and an analyer in between to measure 
tone duration.

We have tried setting chan_dahdi.conf parameter 'toneduration', but that
does not do anything that we can measure.
We have also tried setting channel.c parameter #define 
AST_DEFAULT_EMULATE_DTMF_DURATION 200
to several different values but none seem to alter DTMF duration at all. 

Does anybody have a clue where we can hardcode DTMF duration for tones 
going out of a DAHDI Channel to the PSTN? 

So far we have to tell customer to 'press the buttons' for a little bit 
longer and it will work with the IVRs in question, but many complain 
saying that they don't have to do that with their regular landline or 
cellphone.

Thanks,
Andres
http://www.telesip.net

Andres wrote:
> We are running 1.4.22 and have been experiencing problems with certain 
> IVRs and DTMF Tone duration.  We would like to be able to increase DTMF 
> Tone duration by 50 to 100ms over what the user is pressing on his 
> phone.  We have a PRI test circuit and an analyer in between to measure 
> tone duration.
> 
> We have tried setting chan_dahdi.conf parameter 'toneduration', but
that
> does not do anything that we can measure.
> We have also tried setting channel.c parameter #define 
> AST_DEFAULT_EMULATE_DTMF_DURATION 200
> to several different values but none seem to alter DTMF duration at all. 
> 
> Does anybody have a clue where we can hardcode DTMF duration for tones 
> going out of a DAHDI Channel to the PSTN? 
> 
> So far we have to tell customer to 'press the buttons' for a little
bit
> longer and it will work with the IVRs in question, but many complain 
> saying that they don't have to do that with their regular landline or 
> cellphone.
I suspect you have a DTMF mode mismatch.  If Asterisk is expecting 
RFC2833 or INFO DTMF and the phones are sending inband DTMF then 
Asterisk won't detect it and won't regenerate the DTMF (and so 
toneduration would have no effect).

In Asterisk 1.4 and later there are some DTMF debug options, as well as 
SIP and RTP debug options.  You should start out by making sure that 
Asterisk is detecting the tones as DTMF and not simply passing the raw 
audio thru.

I don't know what specific DTMF and RTP debug commands are in 1.4+ (you 
should be able to look them up in the CLI), as my customers have chosen 
to skip 1.4 and go directly to 1.6 once they have become comfortable 
with it.

Good luck with this.  DTMF issues can be hell to diagnose and fix sometimes.