[Freeswitch-users] Call is interrupted in 30 seconds, configuring NAT on Amazon EC2
stasan89 at gmail.com
Mon Apr 11 19:13:30 MSD 2016
> 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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