atcomsystems.ca/forum Forums TDM Business Telephone Systems Samsung Telephones & Systems Loss of audio when changing to private with public

No whitelist here. Though I did add the public ip address of the remote connection just to be safe.

Those SMT-i5220 IPs are not listed on mmc838, right? If they are, remove them.

Also, please try port forward the below ports:

30000-30031
40000-40031
45000-45031

It does sound like the MGI port's aren't forwarded.

What Temperor says is correct for the 40000 and 45000 range.

MPS/RTG each use 4 ports per channel, you have 8 MPS/RTG channels on a OS7100 so 4 x 8 = 32 you need to goto 4x031.

MGI only uses 2 ports per channel, so 2 x 8 = 16 so 30015 is the end port.

You will also need UDP 6000 forwarded.

You said that MMC838 fixed the issue for the "local" phones, it shouldn't have had any affect on these (unless they are in a different subnet to the system).

You only need to put the remote range into 838 until you change them to register via the public ip address.

The "local" phones are at a remote site but connected over a VLAN so they did have a different subnet. It was a bit strange that they worked fine with nothing in 838 when the system was set to private, but audio conked out when we changed it to priveate w. public.

Not at all.

Since 4.4x software when you have private with public enabled the system assumes anything outside of the subnet of the MP/MGI cards has to use the public address.

Private only has no public ip address so doesn't assume this.

You override this behavior with MMC838. This also applies to SPNET and SIP Trunks in some circumstances.

Prior to 4.4x you could manually select in the ip phone/spnet config whether it was private or public.

I don't know the reason for the change, but i suspect it has something to-do with MPS.

This question gets asked a bit, I might make a sticky.

I forgot to check in. That did the trick nicely.

Now I'm having issues with the same system (updated to 9.5) and remote IP phones.

I can connect to the server, make calls using the sip trunks, but when I call or intercom to one of the digital phones on the main site, there is no audio.

Is it necessary to set forwarding on the network at the remote site at all?

-W

It sounds like an MGI issue.
You do have the MGI ports forwarded?

IP phones to sip trunks will use MPS.

No, you don't need to forward on the remote network.

Yes, the MGI ports are forwarded.
I ran a wireshark on the phone side and it had a few entries that showed the phone sending requests to the local IP of the OS and the info looked like 9000-->40018 LEN=32

How are these phones connecting to the system, using public ip or vpn from site to site?

Biztel: we are using a Public ip.

Now just for kicks and giggles I placed a newly factory reset router on a raw Internet connection and it all worked a treat. So it should follow that the issue is with the router on the outgoing side. Is no ports are specifically blocked so, any idea what might be stopping it?