[Freeswitch-users] Getting DTMF to work with SIPP

[Ned Ludd]

Ken, do you have any examples of how I might do this?

Thanks

On Thu, Sep 8, 2016 at 9:00 PM, Ken Rice <krice at freeswitch.org> wrote:

> Be careful calling 2833 in band as it is not in band dtmf signaling. In
> band specifically refers to the dtmf tones in the actual audio stream. 2833
> is specifically carried on effectively a sub channel in the rtp stream.
> That being said sipp can play rfc2833 at specific times. It can also replay
> rtp atreams or reflect them.
>
> Better method of testing just voicemail may be actually scripting up a
> call generator with freeswitch and some lua. Want to get really fancy add
> some speech rec i there and know of the prompts are playing correctly
>
> Sent from my iPhone
>
> > On Sep 8, 2016, at 7:23 PM, Guillermo Ruiz Camauer <grcamauer at gmail.com>
> wrote:
> >
> > RFC2833 is actually sent in band, but as a special packet in the RTP
> stream.  SIP-INFO is the only real out of band signaling for DTMF, as it
> goes as a SIP packet.
> >
> > Guillermo
> >
> > Sent from my iPhone
> >
> >> On Sep 8, 2016, at 17:49, Ned Ludd <edwardludd at gmail.com> wrote:
> >>
> >> Trying to test FS voicemail with sipp.
> >>
> >> I can't get DTMF to work no matter which way I play it.  Any tips?
> >>
> >> If I use "start_dtmf" in my dialplan, and send DTMF with "play_dtmf" in
> my sipp scenario they should go in-band, right?
> >>
> >> And likewise if I take out "start_dtmf" and use "play_pcap_audio" with
> the 2833 pcap files in my scenario it should go out of band, right?
> >>
> >> Neither work.
> >>
> >> Attaching my scenario file.  Maybe I'm just not doing it right?
> >>
> >> -----
> >> Ned Ludd
> >> <pickup.xml>
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-- 
-----
Ned Ludd
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