[Freeswitch-users] Call is interrupted in 30 seconds, configuring NAT on Amazon EC2

Jurijs Ivolga jurijs.ivolga at gmail.com
Mon Apr 11 19:19:46 MSD 2016


Hi Stas,

You are welcome.

Keep in mind, that in your set-up freeswitch should send back BYE to
internal ip(172.31.0.16), it may even work with external IP(52.*.*.177),
but it will be incorrect set-up...

With kind regards,

Jurijs

On Mon, Apr 11, 2016 at 6:13 PM, Стас Тельнов <stasan89 at gmail.com> wrote:

> Hi Jurijs,
>
>> I assume:
>> UA => OpenSIPS => Freeswitch => VoIP provider
>>
>> 172.31.0.169:5060 - OpenSIPS
>> 172.31.22.124:5060 - Freeswitch
>> 178.*.*.12:5060 -  VoIP provider
>>
>
> Yes, it is correct.
>
> As you can see BYE is strange too... Bye should be sent back to OpenSIPS, but not to 52.*.*.177:5060.
>>
>> It is external sip IP. 172.31.0.16 <http://172.31.0.169:5060> - it is IP
> in amazon local network; 52.*.*.177==sip0.MY_SIP_DOMAIN.com - it is
> public IP and domain of opensips.
>
> Trouble was in opensips ports configures. On clients available only 5061
> (tls) port and disablied 5060 port. In opensips server enabled 5060 and
> 5061 ports and opensips send request to freeswitch by 5060 port.
> Aftrer opensips send 200 OK command and ACK to client by 5060 port, but on
> clients available only 5061 port and packets dont delivered.
>
> Thanks for help, a programmer who does not have experience in servers
> administration and VOIP-telephony is not easy to quickly deal with the
> fact that there is. But with your help, I managed to make it work.
>
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