[asterisk-users] SIP trunk between asterisk boxes

Jerry Geis jerry.geis at gmail.com
Tue Jul 23 11:46:08 CDT 2019


I have a sip trunk between two asterisk boxes.
I can call into the first box, hit 499 for example and the call goes to the
second box and answers as expected plays me audio message just fine etc...
My issue is that DTMF does not seem to be working.

Both sides are set for:
dtmfmode=RFC2833

What might I look at as to why DTMF digits are not transferred?
Thanks,

Jerry
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