[asterisk-users] Media flow between them

[Jerry Geis]

I have a hosted server.
I have TWO different locations what have phones. Chicago and Indiana
If I send audio direct from server to Chicago I hear it - same with indiana.
But if indiana calls chicago - NO AUDIO.

I see this in the  CLI


  -- Channel SIP/63009-00000013 joined 'simple_bridge' basic-bridge
<475050e7-9d99-43f0-a9bf-7aa581a97fd9>
    -- Channel SIP/63000-00000012 joined 'simple_bridge' basic-bridge
<475050e7-9d99-43f0-a9bf-7aa581a97fd9>
       > Bridge 475050e7-9d99-43f0-a9bf-7aa581a97fd9: switching from
simple_bridge technology to native_rtp
       > Remotely bridged 'SIP/63000-00000012' and 'SIP/63009-00000013' -
media will flow directly between them

I added in general section of sip.conf (chan_sip in use)
directrtpsetup=no
directmedia=no

but yet I still see "media will flow directly between them".
HOW do I turn this off - RTP has to go through the server.


Thanks

Jerry
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