[ILUG] Asterisk and DTMF recognition

[Barry Flanagan]

Niall O Broin wrote:
> On 14 Jun 2006, at 08:00, Kevin Brennan wrote:
> 
>> Niall O Broin wrote:
>>> I recently setup an Asterisk server which is the first one I have
>>> setup using an analog line card (a cheap clone). Works nicely
>>> (ignoring the fact that a Grandstream phone refuses to register to
>>> it, but that's another issue) but it seems to have issues recognising
>>> DTMF tones to navigate menus - sometimes it will, sometimes it won't.
>>> Does anyone have any idea what that might be?
> 
>> There are 3 different ways to send DTMF via SIP; 1)in audio 2)RFC 2833
>> and 3) via SIP info. You can normally set this on your phone and with
>> dftmfmode on Asterisk in sip.conf.
> 
> Sorry - I wasn't clear enough as to what the problem was in my original
> post. It's not an issue of how to send DTMF in SIP (I imagine that would
> be a go / no-go situation anyway, and not lead to partial recognition).
> My problem was with recognition of tones on an incoming analog call from
> the PSTN to the Asterisk box via the FXO card.
> 

Here is my /etc/asterisk/zapata.conf which worked with DTMF on a clone
card. Might help.

language=en
busydetect=yes
faxdetect=both
busycount=7
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
useincomingcalleridonzaptransfer=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
group=1
signalling=fxs_ks
callerid=asreceived
context=pstn-incoming ; Points to the default context of your
extensions.conf
channel => 1

-- 

-Barry Flanagan