[asterisk-users] Weird 'hairpin' call rtp audio problem
benoit.panizzon at imp.ch
Fri Aug 10 09:10:20 CDT 2018
> > The "rtp_keepalive" option can be used to have the RTP stack send an
> > RTP packet out. Try that and see what happens.
> Once again 'bullseye' that fixed the problem. Thank you!
Now a customer with and FreePBX 2.9.0 (Asterisk 22.214.171.124) ran into the
same issue with our SBC.
I told him to set rtpkeepalive=1 in sip.conf but I don't see this
version sending any comfort noise packets.
Isn't there any way to disable this nat detection feature completely
in asterisk? (nat=no does not seem to do the trick)
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