SIP ALG problem (stupid plain NAT)
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Hi,
I have an asterisk server sitting behind pfsense, I have successfully configured pfsense for NAT traversal, and clients from the internet can call clients inside the network of asterisk, as well as other clients registered with this asterisk server on the internet.
The problem now is when a client from the internet do a call, the call disconnects in 10~20 seconds, but during this period the call goes fine and voice is heard on both ends; But when a client on the same network of asterisk calls another client registered from the internet, the call is established without any issues, and it doesn't disconnect.
I have also noticed that when internet clients do calls, and the call is established on both ends, if one of the two parties hang up, the other end isn't notified and the call stays opened at this end.
I have asked about this issue on asterisk mailing list and I was told to configure pfsense to do stupid plain NAT and dont touch the sip headers.
How do I do that in pfsense?
Any advice will be very much appreciated
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Maybe you can find a hint here.
http://doc.pfsense.org/index.php?title=Special%3ASearch&search=voip&go= -
I already went through all of them and tried them all.
Setting the static port helped me in "no sound in both calling ends" issue, but they were not helpful in fixing the disconnection issue.
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What about point 2 here?
http://doc.pfsense.org/index.php/VoIP_Configuration
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I already set "Firewall Optimization Options" to "Conservative", but it didn't help
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can you get a sniffer trace and see why (using wireshark) the connection is being torn down?
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The issue was due to a router in the middle of the VoIP cloud which had the option SIP ALG enabled. I just disabled this option and everything worked without any issues.
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Cool.