one way audio on pbx behind nat when changing default sip port

hi,

i have a asterisk server behind mikrotik router . i want to register phones from public ip and a port different from 5060,
when i use 5060 and nat from public ip 37.X.X.X to my internal ip 192.168.2.5 i can register phone from outside and call is established correctly. but when i change nat public port for example 5065 to inernal 5060 i have one way audio issue.

*rtp ports 10000-20000 are nat ed
*mikrotik sip alg disabled

thank for helps

You can use wireshark and capture the VOIP packets, then you can see the analysis and get more details…
If the problem was with the sip port, you would not be able to register the extension..
But the one way audio is related to your rtp ports…

thanks for reply

i dont change rtp port config and nat rules for them.
after that i checked connections in mikrotik firewall and saw that after registration in addition to established connection to 5065 client still try to connect to 5060!.
Is this a problem with the pbx setting?