[Asterisk-Users] What does RELAXDTMF do?

[Doug Crompton]

Peter,

 Perhaps you have not followed the thread over the last few days about
DTMF feedthru??? Here is what I sent out to another list kind of summing
it up....

Regarding DTMF pass thru problems when using the SPA-3000 and *. The
problem manifests itself as the inability to pass DTMF over the FXO to a
PSTN call once the call is established. This would be used to call a bank,
external voicemail or other service and use DTMF signaling to their
service.

To make a long story short (you can go thru the * mailist archives) this
is an * problem in RFC-8233. It has been known for awhile and is being
worked on in the form of a total RFC-8233 rewrite coming in 1.4 *
hopefully this summer. Until then here is the fix I came up with.

The FXO port Sipura setup (PSTN) should be set to INBAND for dtmf and the
codec limited to g711u (or a), on the * side in sip.config FXO context
set dtmf=inband and limit the codec to only g711u (or a)

When you call yourself (say using your cell) and listen on the opposing
phone hitting a key one listening on the other you should hear at least a
half second or so of audible tone. Check this before and after changing
these settings. Using RFC-8233 all I heard was a click and little or no
audible tone.

One other thing is that you CANNOT use features via tones over the FXO
(TtWw,etc flags in dial). This is another broken issue in *. When you
listen over the phone and hit a lead in character, defined in
features.config, * mutes that character and it never gets sent. The
correect action should be that it should mute it and wait until the second
character. If the second character is not sent in a defined time then send
the first character. This is not happenning. This might be an INBAND issue
though and once RFC-8233 is fixed and can be used it might then work.

If you have no need to send DTMF on a connected call via FXO then this
change is not needed and you can use the current RFC-8233 as well as
features. Just remember when you try to send DTMF over FXO port to PSTN
that you know why it does not work!!

This problem was/has been blamed on Sipura but is really an admitted *
problem. It exists with other (but certainly not all) fxo devices
also.

As I said the best way to troubleshoot this is to actually call yourself
and listen. Otherwise you are shooting in the dark and guessing.

Doug


On Thu, 8 Jun 2006, Peter J Dean wrote:

> I have an issue with DTMF. DTMF is being partly recognised by some
> external IVR systems (banks, billing, etc), other IVR systems have
> intermittent issues. Call our VSP directly and using their IVR system
> without issue, and our internal IVR works just fine. Currently i have
> all voip devices using RFC2833, which is what is recommended, and
> thus the sip.conf file has dtmfmode=rfc2833 and relaxdtmf=yes.
>
> I have not seen any information that clearly defines the purpose of
> the relaxdtmf parameter in the sip.conf file, and wondering of
> flicking it from yes to no will have an impact, and if so what sort
> of impact will it have?
>
> Redhat FC4 + updates
> Asterisk v1.2.9.1
> SNOM v6.0.3 beta
> SPA3000 v3.1.10d
>
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****************************
*  Doug Crompton	   *
*  Richboro, PA 18954	   *
*  215-431-6307		   *
*		  	   *
* doug at crompton.com        *
* http://www.crompton.com  *
****************************