asterisk users - Oct 2022 - Trying asterisk on AWS

I am trying to get audio to work on AWS using asterisk 18.14.0

I have enabled the firewall to allow ALL UDP on AWS

My SIP extension has
nat=force_rport,comedia
qualify=yes
allow=ulaw
allow=alaw
allow=gsm
canreinvite=yes

I enable "rtp set debug on" and the console is printing info.

The call comes into my linphone softphone - but I get no audio on my
linphone softphone.
What might I be missing to allow the audio ?
Volume is up.

Thanks

Jerry
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On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis <jerry.geis at gmail.com> wrote:
> I am trying to get audio to work on AWS using asterisk 18.14.0
>
> I have enabled the firewall to allow ALL UDP on AWS
>
> My SIP extension has
> nat=force_rport,comedia
> qualify=yes
> allow=ulaw
> allow=alaw
> allow=gsm
> canreinvite=yes
>
> I enable "rtp set debug on" and the console is printing info.
>
> The call comes into my linphone softphone - but I get no audio on my
> linphone softphone.
> What might I be missing to allow the audio ?
> Volume is up.
>
> Thanks
>
> Jerry
>

I just noticed the RTP log is sending to 192.168.2.0 which is my local lan
address of the linphone - it should be sending to the NAT address and is
not.
What did I not set correctly ?
I am not using pjsip - but the older asterisk.

Thanks

Jerry
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