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Asterisk 服务器的线路呼叫
原标题:Route Calls in Asterisk Server
I have an Asterisk server with a public IP address (without using NAT). Additionally, I have two IP phones behind different NATs, each with a different mapped public address. Both phones can register on my Asterisk server, but when one of the IP phones calls the other, I observe a loop of received INVITE packets on Wireshark on the Asterisk server without seeing the TRYING and RINGING packets. As a result, the SIP session never establishes. my SIP.conf [general] context=default allowguest=no bindport=5060 bindaddr=0.0.0.0 allow=ulaw allow=alaw externip=105.96.25.87 localnet=192.168.1.0/24 nat=force_rport,comedia [148] type=friend context=default host=dynamic nat=force_rport,comedia directmedia=no [151] type=friend context=default host=dynamic nat=force_rport,comedia directmedia=no [170] type=friend context=default host=dynamic nat=force_rport,comedia directmedia=no I want a solution to the problem I mentioned above.
问题回答
We not see your trace so no way to say anything exactly. But likely you have incorrect NAT mapping or phone s NAT is firewalled and not accept connections.




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