Ok, Who knows what's going here...Here is my Scenario..
We're looking at a new Phone Platform and I'm only able to get a NAT to work part of the time. First, when the IP Phone loads, internal address of 172.23.1.1, It connects out to the Platform IP of 184.108.40.206, downloads the config from it's TFTP Service. Since we don't want our Voice traffic mingled in with our other public traffic, we've NAT'd 172.23.0.0 /16 to a Public IP of 220.127.116.11, the NAT Works perfectly here. Next, once the TFTP Load is complete, the Phone tries to register via SIP, same Internal Address 172.23.1.1 to the platform of 18.104.22.168, however the same NAT Statement is not being used. I've ran numerous PCAP's, changed NAT several times, moved rules, but everything appears correct. Furthermore, when I run test nat-policy-match with the proper destination and source on port 5060 and protocol 17, it test out correctly.
Inside Zone to Destination Zone, source address 172.23.0.0/16 to destination address 22.214.171.124, any interface and any service translate type Dynamic IP address of 126.96.36.199. Anyone have any idea what's going on and why the NAT isn't getting applied 100% of the time? When I look at the Web GUI of the phone platform, it's showing my phone as being registered with our Public IP of 188.8.131.52. I am able to place and receive a call, however there is not audio. Running another PCAP for a call session, all UDP packets from 172.23.1.1 are getting dropped.
One thing to note, SIP ALG is turned enabled, though I'm not sure if that's the issue. Do try to get around SIP ALG, I created a custom Application, mirrored from SIP, and setup an Application Override with Inside/Outside, all IP's and UDP 5060, but still having the issue.
I've opened a Support Case, but with it being a low priority, thought I'd reach out to the community to see if anyone has ran into this issue inthe past.
We're running 8.0.8 PAN-OS version.
Going through the logs, nothing is being blocked by policy to/from by private address/NAT'd Address and the PBX. Actually, the policy is set to allow any application, Service is Application-Default. I enabled logging on my Interzone rule, however nothing is being dropped by policy, that I can find.
I hear ya it can be a pain. We use Skype and I can tell you that even having the application set to any and service to application default was not enough. Skype/Lync use like almost every port 50K and up :(. Does your SIP terminate internally or does it flow out over the internet to the provider? I would also say call the provider and see if they can run a trace while your making the test call just to see if they see anything on their end.
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