[Freeswitch-users] Getting DTMF to work with SIPP

[Ken Rice]

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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