Ring group issue

[MikeyG.IT]

I have a Google cloud instance, all extensions are provisioned using remote stun. Using yealink t27gs, all with most recent firmware. I am having an issue where not every extension part of the incoming call ring group rings. A call will come in and every so often when the call rings, not all extensions will ring when they are all part of the same ring group with the ring all strategy. There have been no changes to the phone system that I am aware of and this problem just seemed to start happening on its own.

 

[leejor]

Area all sets at the same location? Check the 3CX Activity Log to be certain that Invites were sent to all extensions, on one of the failed calls. It may be, that, if all invites are sent to the same public IP (different ports), that one (or more) are getting lost along the way, at times. You would have to confirm with Wireshark, at the remote end, that all Invites make it to the remote LAN.

 

[Saqqara]

How many phones are the remote site ? You maybe better off using SBC or site to site vpn where you config the phones as local.

If you have a few phones, have you given each phone different ports and setup port forwarding https://www.3cx.com/community/threa...etgear-nighthawk-x4s-d7800.54620/#post-225296

 

[MikeyG.IT]

Interesting. I have deployed this scenario on more that one occasion and have never forwarded unique sip ports to each extension. So from that thread, it is saying to forward a different SIP port to each remote extension?
Extension A - SIP port 5060
Extension B - SIP port 5061
Extension C - SIP port 5062
And so on?

Would there need to be any port forwarding done on the Google hosted VM other than whats created when initially configuring the system with the PBX express?

 

[cobaltit]

Not just the SIP port but the RTP ports as well.

https://www.3cx.com/docs/manual/configuring-ip-phones/#h.ul2fzupi6t22/

Check the bottom of this document.

 

[MikeyG.IT]

In that article it says you can leave the sip port default and then says later in the article that it could cause audio issues. My issues is that sometimes a phone in a ring group wont ring at all.
I have other systems I manage that are configured this same way and don't have issues and the issue started happening long after it was implemented so I have been scratching my head trying to figure it out. Of course its intermittent which makes it even more fun to troubleshoot.

 

[leejor]

If you have multiple sets, at one location, using STUN, each is seen by the PBX as (PublicIP) +(port). So to properly address each, they have to be unique. In most cases, a single set, or two (sometimes) can get away with no port forwarding set-up at the remote end, but, how well this works this can depend on the particular router used there.

 

[MikeyG.IT]

Ok gotcha. A little more info on the setup, the remote site is using a Unifi USG Pro (gateway).
Will there need to be any additional port forwarding done on the Google VM that hosts the PBX?
Also, if I use a SBC with that eliminate the need for the unique sip and rtp ports?

I greatly appreciate everyones help!!

 

[leejor]

Will there need to be any additional port forwarding done on the Google VM that hosts the PBX?

No, as long as the Firewall Checker passes all, that end should be ok.

 

[MikeyG.IT]

Ok, to confirm:
In the 3CX dashboard, go to all the extensions and assign them unique SIP and RTP ports in the phone provisioning tab
Forward those ports to each extension on the gateway
Ext. 1 sip 5060
Ext. 2 sip 5061
And so on, and also the RTP ports as well?

 

[leejor]

Forward those ports to each extension on the gateway

If by "gateway", you mean the router, at the remote end, then, yes.

 

[cobaltit]

In that article it says you can leave the sip port default and then says later in the article that it could cause audio issues. My issues is that sometimes a phone in a ring group wont ring at all.
I have other systems I manage that are configured this same way and don't have issues and the issue started happening long after it was implemented so I have been scratching my head trying to figure it out. Of course its intermittent which makes it even more fun to troubleshoot.

Yes for the setup of a single phone you can leave the SIP port at default. But the bottom section that I referenced says this:

1. If you have multiple IP Phones on the same remote network configured with the same SIP and RTP ports, you might have an audio problem caused by the way certain routers implement NAT. In this scenario, each phone must have a different SIP Port and a range of RTP ports must be configured per phone. To do this, click on the Provisioning tab, change the SIP and RTP ports and perform a re-provision of each phone from the Phones node.

Ok gotcha. A little more info on the setup, the remote site is using a Unifi USG Pro (gateway).
Will there need to be any additional port forwarding done on the Google VM that hosts the PBX?
Also, if I use a SBC with that eliminate the need for the unique sip and rtp ports?

I greatly appreciate everyones help!!

Yes

https://www.3cx.com/docs/3cx-tunnel-session-border-controller/