Hi,
I’m migrating my asterisk server to cloud but facing few issues. One of which is No voice issue when I turn the IPtables ON. Neither can I send any DTMF (dial any extension). If I turn the IPtables OFF, I have two way audio.
How can I troubleshoot the issue, I dont have much experience with Asterisk or linux
Assuming SIP.
Assuming chan_sip.
Check rtp.conf and make sure that you have firewall pass rules for the port range specified there.
Make sure you have firewall pass rules for any destination port outbound. In practice you could restrict the source port to the above range.
If you still have rouble, use “sip set debug on” and look at the SDP to confirm which port numbers are actually in use, but remember the peer gets to choose the number at its side.
Actually i changed the bind port to 5050. I believe I have to allow it in IPTables. Its UDP 5050 correct?
iptables -I INPUT -p udp -s xxx.xxx.xxx.xxx/32 --dport 5050 -j ACCEPT
where xxx.xxx.xxx.xxx is my server IP
That is only for signalling. You also have to open the RTP ones, for the speech.
For RTP
iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT