Conference to add a new caller results in no audio

[seantp]

So I've got a remote worker and I set him up with the Asus AC1900 router and I have the RT-AC68U

I have SIP ALG disabled on both routers.
On the phones, I usually switch STUN to Disabled under NAT because I had issues with clients
Running latest version of 3CX.

To test I'll call my cell phone, hit the conference button and then dial the extension of my remote worker. It rings he picks up but we cannot hear each other. Same when he tries it.

I tried every combination, SIP ALG on & OFF along with NAT STUN/ Disabled. Nothing works. Not sure where to look next.

 

[mariosM_3CX]

Hello @seantp

What phone/clients are involved in the call and how are they provisioned?
Do you have the same issue when calling the remote worker directly instead of using conference?

 

[seantp]

Calling the remote worker directly works fine. It's only when I'm on a call and try to conference him in. He has a yealink t46s provisioned by 3cx while I have a t48g.

 

[mariosM_3CX]

You may need a capture to see exactly what is happening with the audio.
Is the remote phone behind an SBC or is it remote stun? In case of remote stun, make sure that you port forward SIP and RTP ports on the phone.
Can you also enable the option "PBX Delivers Audio" in the extensions settings and retry?

 

[seantp]

The 3CX server is in the cloud. We are both home offices and we're using STUN. No SBC in the mix.

However, PBX Delivers Audio was not checked. After enabling that on all the extensions it worked. One thing I'm not doing is forwarding RTP ports manually to the phone. How important is that, as it seems things are working. Is it possible the Asus router is routing the signals correctly without the port forwards?

 

[mariosM_3CX]

Even though (and evidently) this could work, you are leaving things uncontrolled and put your faith on the router to correctly route the traffic. Some routers may deal with this better than others but safest way will be to do the forwarding.