Hi,
I am having problems with one way audio on my freepbx. So there is a outgoing audio but not incoming on calls.
There is one sip trunk. I am using sophtpones on local network 192.168.100.0/24. Also one interface on freepbx is in that network. Sip interface is on 10.79.104.0/22.
There is a router with external address 217.71.54.244.
So ive tryed several things, forwarding rtp ports, changing nat settings , etc.
After sending logs i recieved info that registration of the sip is coming from 10.xxxx intreface but media is coming from external address 217.71.54.244.
So there is a routing problem but i have no idea what to do.
Also i got info that media for calls is coming from
Sounds like you’ve not enumerated all local networks/subnets in Settings → Asterisk SIP Settings. Changes on this page may require an Asterisk restart. [ edit - missed david55’s reply before replying]
All network setting are ok, i cant find alg sip on huawei ont.
Tryed zilion settings but same thing. One way audio persists, and call disconnects after 30 sec