[asterisk-users] Using Sangoma Call Progress Analysis behind NAT router

[Jim Dickenson]

I work at home with standard residential cable Internet service and I wanted to test CPA for use with our dialer solution. The first problem I ran into is that CPA only works with a SIP provider that does IP based authentication opposed to usename/password authentication. After I got an account setup to solve that problem I thought I was on my way to being able to test.

No so.

I got asterisk making outbound calls via the SIP provider.

I got CPA installed and ready to go.

I made a test call and although the call gets setup and I can hear audio in both directions CPA did not have any audio to analyze. I then looked at both the SIP messages asterisk sent and received as well as the SIP messages that CPA sent and received and saw that the invite message has the internal IP address for where RTP traffic is to be sent in the CPA messages.

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This diagram from CPA's user manual sure looks like my setup. I contacted Sangoma support and they say the product does not do NAT transversal.

Has anyone found a work around this problem?
-- 
Jim Dickenson
mailto:dickenson at cfmc.com

CfMC
http://www.cfmc.com/