YUP, this is the way that asterisk works. It is going to quelch all DTMF that
goes out via a SIP gateway via asterisk.
I spent a long time working this through and it has to do with the way that
asterisk deals with DTMF and the DSP.c module that
sits inband to the RTP/audio stream. There is a flag called
DSP_DIGITMODE_NOQUELCH that is broken that might allow the inband
DTMF after answer to work on inband but its broke.
If you do not use asterisk as your gate/ to/from the PSTN you are going to have
a issue with DTMF after connect. There are a couple
of kludges that can get it to work part of the time. But from my experiance DTMF
is not handled correctly in asterisk if you use any
gateway other then asterisk.
IE: you use a cisco or TNT as your gateway to/from the PSTN via SIP and asterisk
to talk to the 2500 type phones.
-larry
> Message: 1
> Date: Thu, 16 Mar 2006 12:36:45 -0800
> From: Martin Joseph <ast@stillnewt.org>
> Subject: [Asterisk-Users] RFC 2833 and SIP? DTMF? What am I not
> getting?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Message-ID: <5ebf9b207fd5e5f2fc8dab598d4f24f6@stillnewt.org>
> Content-Type: text/plain; charset=US-ASCII; format=flowed
>
> Hi again,
>
> I am trying to get my DTMF to use RFC 2833 rather then inband, so
> that
> I can utilize lower bandwidth codecs through my FXO.
>
> After much tinkering I was able to get my gateway (wellgate 3701A)
> configured to a point where I have some success, but no real joy.
>
> I have configured the RTP Payload type (or RFC2833 Payload type) to
> 101. I don't have a clue what this means, but I took the 101 from
> my
> AG168V ATA's configuration screen, as I know that device seemed to
> work
> fine through the old HT-488 fxo(via rfc2833).
>
> I then changed my asterisk extensions for both the FXS and FXO on the
> wellgate to include dtmfmode=rfc2833.
>
> This has brought me to a point where both my hardphones (ATA's) and
> my
> softphones (IAXcomm, or JackenIAX) work perfectly with comedian mail.
>
> To me this means that asterisk is properly getting the RFC2833 events
> from the user agents.
>
> BUT, if I try to dial out the FXO, none of my phones (hard or soft)
> produce working touchtones for a PSTN based voicemail system.
>
> Even stranger to me, is the fact that from the phone connected to the
> FXS on the wellgate I can hear tones(listening on a called line), but
> they sound kind "rough" at the edges. From the AG168V I hear no
> tones, but what seems to be "blown out" tones (ie overdriven
> volume).
> From the IAX softphones I hear no tones at all just clicks!
>
> Now I would have guessed that the FXO would be doing the conversion
> of
> the RFC2833 to inband, so that I thought all the tones should sound
> the
> same from any phone? Apparently this isn't the case at all.
>
> Thanks to all of you for any help understanding and or debugging this
> mess.
>
> Marty
>
> PS I spent a good deal of time adjusting the DTMF volume for the
> wellgate FXS/FXO hoping this might help before I discovered the
> variety
> of non working DTMF being generated.
>
>
>
> ------------------------------
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