[Asterisk-Users] DTMF issues

Greg Hill gregh-asterisk at hillnet.us
Tue Aug 10 12:23:27 MST 2004


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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