Thanks for the information
This is now working...
externip=EC2 public IP
localnet=EC2 local range
nat=force_rport,comedia
I got audio, Fantastic
Jerry
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -
On Thursday 06 October 2022 at 15:24:22, Jerry Geis wrote:
> I added:
>
> externip=xxx
> nat=force_rport,comedia
>
> to the general section of sip.conf
>
> its still sending to the local IP.
Does your local router (the one connecting Linphone to the Internet) have a
"SIP helper" or "SIP ALG"
On Thu, Oct 6, 2022 at 10:24 AM Jerry Geis wrote:
> >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)
>
> I
>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)
I added:
externip=xxx
nat=force_rport,comedia
to the general s
On Thu, Oct 6, 2022 at 10:17 AM Jerry Geis wrote:
>
>
> On Thu, Oct 6, 2022 at 9:02 AM Jerry Geis wrote:
>
>>
>>
>> On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis 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
On Thu, Oct 6, 2022 at 9:02 AM Jerry Geis wrote:
>
>
> On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis 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=y
On Thu, Oct 6, 2022 at 10:03 AM Jerry Geis wrote:
>
>
> On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis 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=
On Thu, Oct 6, 2022 at 8:59 AM Jerry Geis 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
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 come