[asterisk-users] Media flow between them

[Jerry Geis]

On Thu, Jul 20, 2023 at 10:24 AM Jerry Geis <jerry.geis at gmail.com> wrote:

> 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
>

even easier:
canreinvite=no
I had yes.

works

Jerry
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