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skeates

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I've just moved our Kerio Operator into a hosted environment. The only differences between how it is setup now and was before the move is that now it is not local to our network and the firewall is now Kerio Control.

All appears to be working incoming and outgoing calls are working the only issue is I am not getting any audio either way. I can't even get voicemail or music audio by dialling 82.

Usally when this happens its due to ports not being open usually the RTP 10000 - 20000 and when I added the firewall rule in to Control it did not include these ports so I had add a new service and add them to the rule. This however still appears not to be working.

I'm fairly familiar wth Connect and Operator. I've just started using Control so I might have missed something in the firewall.

Any help would be appreciated.
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nhoague

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1) make sure Operator has NAT enabled and the correct external IP address under the Network tab
2) make sure your extensions have NAT enabled. I always do this regardless since NAT and VoIP aren't friends. Smile

As for Control, I usually forward all Kerio operator service group and it works well.

Depending on your ISP, we have Comcast, we have to turn off the built in firewall on their modem too.

Good luck!
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skeates

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Thanks for the reply.

Under networking I have NAT enabled. Public IP is being detected correctly. I've changed to something else asked it to update and it has done so.

Under extensions each extension is ticked as Extension is behind NAT.

The server is in a data centre with no firewall in front of it which is why we are using Control.

Do you need to create any outbound Traffic Rules or only need the inbound?
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nhoague

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Ok good info.

I have also created outbound firewall rules assigning the outbound IP correctly. This in my case was because our ISP comes in via a different IP that goes out. You may or may not have that problem.

Our SIP Prodivder also requires a ACL on the IP address so I have had to register that with them.

Are you able to get the ring tone when dialing a number? The SIP has two parts. One for the session initiation which actually causes the phone to ring and the second for audio. Assuming you have the picture working, then audio should come down to a NAT or a redirected IP somewhere.

Sorry I'm kinda rambling just blurting out my thoughts. Smile
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skeates

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I've got the address of the firewall (control) as .40 operator is on .42 so I set a rule saying outbound traffic for operator go out on .42 and another one saying inbound traffic on to .42 direct to operators internal address.

I'm getting a dial tone when calling out and the phone on the other end rings, but again still no audio. It's doing my head in. I'm pretty sure it's to do with Control settings. When I had the same setup previously using a draytek I just did the port forwards and told the draytek to route all outbound traffic for a specific machine via the required address all worked fine.

Think I'll bash my head a bit longer and if no luck give Kerio support a call tomorrow.
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skeates

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Managed to sort the problem. The RTP ports I had defined in services I had set the source ports as 10000-20000 and the destination ports as 10000-20000

When I was clicking through the menus I notices all the other services had been defined as source port any so changed it and it started working.
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nhoague

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Ahh good to hear! I didn't even think of the source as your problem. :/
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