[asterisk-users] Trying asterisk on AWS

[Jerry Geis]

On Thu, Oct 6, 2022 at 9:02 AM Jerry Geis <jerry.geis at gmail.com> wrote:

>
>
> 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
>

 >Have you configured chan_sip to know it is behind NAT itself and what its
>public IP address is? If not, then you'll get no audio.

I'm thinking I have not. What did I miss ?

Thanks,

Jerry
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