[asterisk-users] SIP DTMF Troubleshoot

[Andrew Joakimsen]

Too much info then too little info.

Basically the issue is the provider this happens even when we send
them the calls in IAX because they talk SIP to the same gateway.

I just need to prove it to these people. Anyone have any DTMF issues
between Asterisk and a Quintum gateway?

On Jan 28, 2008 6:47 PM, Jared Smith <jsmith at digium.com> wrote:
>
> On Mon, 2008-01-28 at 18:05 -0500, Andrew Joakimsen wrote:
> > How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
> > messages related to DTMF... or if I just do a global SIP debug for
> > that matter.... I am using RFC DTMF but it's not being passed to the
> > PSTN and I need to debug this further. I've tried to increase the
> > verbosity and the debug ('set debug n') and that didn't help either. I
> > assume this is because even RFC2833 sends the DTMF as RTP which
> > wouldn't show up anyways.... but how to troubleshoot DTMF issues?
>
> I'd first turn on "rtp debug" and see if that helps.  If that's not
> enough information, I'd go into logger.conf and add "dtmf" to the logger
> and messages lines (and any others you care about), and then do a
> "logger reload" from the Asterisk CLI.
>
> --
> Jared Smith
> Community Relations Manager
> Digium, Inc.
>
>
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