[Asterisk-Users] Re: SIP <-> PSTN DTMF

[John Goerzen]

On 2005-03-20, goldhorse <goldhorse at yahoo.co.jp> wrote:
> I use a Handytone 486. While you have to configure dtmfmode=rfc2833 in 
> asterisk,
> it will not work if you do not set the dtmf mode to "SIP info" in the 
> ATA itself.
> So you might try different combinations fo dtmf modes in both asterisk 
> and
> the phone you are using until you get the correct one.

Hi Daniel,

Thanks for the tip; unfortunately, it doesn't help.

I have confirmed correct dtmfmode settings.  It works fine when I call
something such as Asterisk voice mail.  Also, when I turn on SIP
debugging, I can see it recognizing DTMF signals from the VOIP phone.

I have tried this both with the linphone softphone and a SPA-841 phone
and have had the same problem with both.

The SPA-841 phone has a nifty feature where I can change the DTMF
signaling.  I've tried it with INFO and Inband.  In info mode, I of
course set info mode in sip.conf.  Again, Asterisk voicemail works fine.
But there is no DTMF at all on calls that are routed outside to the PSTN
via X100P.

If I change the phone and sip.conf to inband, again, it works fine with
Asterisk voicemail.  When I place a call via the X100P, this time, on
the other end of the connection, I can hear a faint DTMF tone -- usually
corresponding to the beginning and end of the tone I would expect it to
send.

It seems that Asterisk is specifically removing the DTMF tones before
sending them on their way!  I don't get it!

Also, one other note, the X100P phone does the initial dialing just
fine.  Never any problem there.  It's just the post-dial DTMF that's
giving me trouble.

Again, help appreciated!

Thanks,

-- John