Chan_console asterisk 18.18.0

HI -

I am using chan_console and chan_sip. Asterisk 18.18.0
I see the RTP coming in with “rtp set debug on”,
I see the call come in with CLI
I can do system(aplay my_wav_file) and hear it
but I do not hear any audio from the SIP call.


input_device = default ; When configuring an input device and output device,
output_device = default ; use the name that you see when you run the “console
; list available” CLI command. If you say “default”, the
; system default input and output devices will be used.
autoanswer = yes
context = default
extension = s
callerid = Yes <(317) 555-1212>
language = en
overridecontext = no
mohinterpret = default
active = yes

Log shows no errors and connects me to Dial(Console/default)
What can be happening ?
Other computers - same install - work just fine.



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