[asterisk-users] Trying asterisk on AWS
Joshua C. Colp
jcolp at sangoma.com
Thu Oct 6 08:19:19 CDT 2022
On Thu, Oct 6, 2022 at 10:17 AM Jerry Geis <jerry.geis at gmail.com> wrote:
>
>
> 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 ?
>
The sample configuration file outlines how things work, and the options for
it:
https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L874
in general localnet and externip (or externaddr, or externhost)
--
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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