[Asterisk-Users] DTMF issues

[dbruce at bananatel.com]

The problem you are having with DTMF is due to the way that the 5350 handles
DTMF...

Your solution is to modify the dial peer on the 5350 that handles the your
calls so that it uses a compatable format by adding "dtmf relay cisco-nte",
Named Telephone Events. There should be 2 dial peers on the 5350 that you
need to modify, one handles inbound calls from asterisk and the other
handles outbound to asterisk.

You should set the DTMF mode in your SIP or H323 configuration to use
rfc2833.

Be aware that some versions of the cisco IOS for the 5350 do not support
named NTE... versions later that 12.2(2)T should work.

ie:

dial-peer voice 1403 voip
   destination pattern 1403*
   session protocol sipv2
   session target sip-server
   dtmf-relay cisco-nte
!

Regards,
  Derek


----- Original Message -----
From: "Greg Hill" <gregh-asterisk at hillnet.us>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, August 10, 2004 1:23 PM
Subject: RE: [Asterisk-Users] DTMF issues


> On Tue, 10 Aug 2004, AJ Grinnell wrote:
>
> > I hadnt heard of that setting until today either, but it still doesnt
work.
> > I am using dtmfmode-rfc2833 in sip.conf, and I have my spa's set to avt
or
> > auto. The "internal" dtmf used for voicemail or anything within * works
just
> > fine. I hust cant get tones out to the PSTN. The DTMF sounds very
distorted
> > on the other end of the call. I am trying to use rfc2833. If you have
any
> > ideas, please let me know. Thank you.
>
> it wasn't until a few minutes after I posted that I realized I should have
> recommended using 'sip show channel ...' while the call is active.
> Asterisk will tell you which DTMF mode it's trying to use on the channel.
> That's how I initially discovered the setting for Broadvoice users to
> receive dtmf on inbound calls.. I knew it should be inband, but sip show
> channel told me that asterisk was using rfc2833 for that leg of the call.
> I went looking for a way to change that setting and found a solution to
> the problem.
>
> If your codec supports inband, you could give it a try.. (I prefer
> out-of-band too, but if the endpoint only supports inband... then you
> can't beat 'em and end up joining 'em.)
>
> oh, you said "dtmfmode-rfc2833" above. That was a typo, right..? You
> really have a '=' rather than '-' in sip.conf?
>
> as for SIPDtmfMode, try a case-insensitive grep on the source to see if
> that string comes up anywhere. If it isn't there, then it's not a real
> option and would probably be ignored by the config-parsing algorithm.
>
> Greg
>
>
>
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com
> > [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Greg Hill
> > Sent: Tuesday, August 10, 2004 1:32 PM
> > To: Asterisk
> > Subject: Re: [Asterisk-Users] DTMF issues
> >
> >
> > On Tue, 10 Aug 2004, AJ Grinnell wrote:
> >
> > > I am now at a total loss. Using Sipura spa-2000s connected to *, I get
> > > DTMF working just fine for internal extensions, voicemail, etc. If
> > > making an outgoing call like this spa --> * --> Cisco AS5350 --> PSTN,
I
> > > get no dial tone. I am working unsuccessfully with Cisco right now on
> > > this, but they cant find anything wrong. I have tried all suggestions
I
> > > can find from the list and elsewhere. I have added SIPDtmfMode to my
> > > outgoing extensions, that still doesnt help. Does anyone out there
have
> > > experiance or ideas with this setup?
> >
> > I haven't heard of any SIPDtmfMode setting, but there is dtmfmode= (in
> > sip.conf, not extensions.conf). Which dtmf mode are you hoping to use?
>
>
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