[Freeswitch-users] Getting DTMF to work with SIPP

[Ned Ludd]

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160908/be77dbc2/attachment.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: pickup.xml
Type: text/xml
Size: 3887 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160908/be77dbc2/attachment.xml