I’m new to asterisk and i’m desperate for help.
I got a simple enviroment, One linksys spa3000 that act as fxo/PSTN converter and an asterisk 13 server that route 10 phone. No firewall or NAT. When i answer an incoming call i can hear the audio and speak but the One Who call can’t.
Any idea? What i should check? The system and the architecture is a clone of What i did on another enviroment Where everything work as It should.
Sorry for my bad english
Cheers to anybody will help me
Assuming you clone this system, might be posible you didnt update this parameter according to the new enviroment .
externaddr = 12.34.56.78 (your public IP)
localnet=192.168.0.0/255.255.0.0 ; RFC 1918 addresses
localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
On sip devices if there are behind nat
nat=force_rport,comedia
;
Thanks i Will check! Where should i put this parameters? I Only deleted the 70-persistent-net.rules to refresh the MAC